From 71f6e0645be42f93c0f90dfcc93b9d2d277c2ee6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 2 Dec 2009 15:11:08 +0900 Subject: ASoC: sh_fsi: avoid using global variable Current FSI driver use global variable to access device data. But this style will be broken if SuperH come with multiple FSI blocks in future. To solve this problem, this patch use cpu_dai->private_data. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9c49c11..7506ef6 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -92,6 +92,7 @@ struct fsi_priv { void __iomem *base; struct snd_pcm_substream *substream; + struct fsi_master *master; int fifo_max; int chan; @@ -110,8 +111,6 @@ struct fsi_master { struct sh_fsi_platform_info *info; }; -static struct fsi_master *master; - /************************************************************************ @@ -166,7 +165,7 @@ static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); } -static int fsi_master_write(u32 reg, u32 data) +static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -175,7 +174,7 @@ static int fsi_master_write(u32 reg, u32 data) return __fsi_reg_write((u32)(master->base + reg), data); } -static u32 fsi_master_read(u32 reg) +static u32 fsi_master_read(struct fsi_master *master, u32 reg) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -184,7 +183,8 @@ static u32 fsi_master_read(u32 reg) return __fsi_reg_read((u32)(master->base + reg)); } -static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) +static int fsi_master_mask_set(struct fsi_master *master, + u32 reg, u32 mask, u32 data) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -200,43 +200,29 @@ static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) ************************************************************************/ -static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream) +static struct fsi_master *fsi_get_master(struct fsi_priv *fsi) { - struct snd_soc_pcm_runtime *rtd; - struct fsi_priv *fsi = NULL; - - if (!substream || !master) - return NULL; - - rtd = substream->private_data; - switch (rtd->dai->cpu_dai->id) { - case 0: - fsi = &master->fsia; - break; - case 1: - fsi = &master->fsib; - break; - } - - return fsi; + return fsi->master; } static int fsi_is_port_a(struct fsi_priv *fsi) { - /* return - * 1 : port a - * 0 : port b - */ + return fsi->master->base == fsi->base; +} - if (fsi == &master->fsia) - return 1; +static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *machine = rtd->dai; + struct snd_soc_dai *dai = machine->cpu_dai; - return 0; + return dai->private_data; } static u32 fsi_get_info_flags(struct fsi_priv *fsi) { int is_porta = fsi_is_port_a(fsi); + struct fsi_master *master = fsi_get_master(fsi); return is_porta ? master->info->porta_flags : master->info->portb_flags; @@ -314,27 +300,30 @@ static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) { u32 data = fsi_port_ab_io_bit(fsi, is_play); + struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(IMSK, data, data); - fsi_master_mask_set(IEMSK, data, data); + fsi_master_mask_set(master, IMSK, data, data); + fsi_master_mask_set(master, IEMSK, data, data); } static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) { u32 data = fsi_port_ab_io_bit(fsi, is_play); + struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(IMSK, data, 0); - fsi_master_mask_set(IEMSK, data, 0); + fsi_master_mask_set(master, IMSK, data, 0); + fsi_master_mask_set(master, IEMSK, data, 0); } static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) { u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); + struct fsi_master *master = fsi_get_master(fsi); if (enable) - fsi_master_mask_set(CLK_RST, val, val); + fsi_master_mask_set(master, CLK_RST, val, val); else - fsi_master_mask_set(CLK_RST, val, 0); + fsi_master_mask_set(master, CLK_RST, val, 0); } static void fsi_irq_init(struct fsi_priv *fsi, int is_play) @@ -355,23 +344,23 @@ static void fsi_irq_init(struct fsi_priv *fsi, int is_play) fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR); /* clear interrupt factor */ - fsi_master_mask_set(INT_ST, data, 0); + fsi_master_mask_set(fsi_get_master(fsi), INT_ST, data, 0); } -static void fsi_soft_all_reset(void) +static void fsi_soft_all_reset(struct fsi_master *master) { - u32 status = fsi_master_read(SOFT_RST); + u32 status = fsi_master_read(master, SOFT_RST); /* port AB reset */ status &= 0x000000ff; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); mdelay(10); /* soft reset */ status &= 0x000000f0; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); status |= 0x00000001; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); mdelay(10); } @@ -517,12 +506,13 @@ static int fsi_data_pop(struct fsi_priv *fsi) static irqreturn_t fsi_interrupt(int irq, void *data) { - u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; - u32 int_st = fsi_master_read(INT_ST); + struct fsi_master *master = data; + u32 status = fsi_master_read(master, SOFT_RST) & ~0x00000010; + u32 int_st = fsi_master_read(master, INT_ST); /* clear irq status */ - fsi_master_write(SOFT_RST, status); - fsi_master_write(SOFT_RST, status | 0x00000010); + fsi_master_write(master, SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status | 0x00000010); if (int_st & INT_A_OUT) fsi_data_push(&master->fsia); @@ -533,7 +523,7 @@ static irqreturn_t fsi_interrupt(int irq, void *data) if (int_st & INT_B_IN) fsi_data_pop(&master->fsib); - fsi_master_write(INT_ST, 0x0000000); + fsi_master_write(master, INT_ST, 0x0000000); return IRQ_HANDLED; } @@ -548,7 +538,7 @@ static irqreturn_t fsi_interrupt(int irq, void *data) static int fsi_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); const char *msg; u32 flags = fsi_get_info_flags(fsi); u32 fmt; @@ -667,7 +657,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, static void fsi_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; fsi_irq_disable(fsi, is_play); @@ -679,7 +669,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); struct snd_pcm_runtime *runtime = substream->runtime; int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret = 0; @@ -760,7 +750,7 @@ static int fsi_hw_free(struct snd_pcm_substream *substream) static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); long location; location = (fsi->byte_offset - 1); @@ -870,10 +860,16 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); ************************************************************************/ static int fsi_probe(struct platform_device *pdev) { + struct fsi_master *master; struct resource *res; unsigned int irq; int ret; + if (0 != pdev->id) { + dev_err(&pdev->dev, "current fsi support id 0 only now\n"); + return -ENODEV; + } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); irq = platform_get_irq(pdev, 0); if (!res || !irq) { @@ -899,15 +895,19 @@ static int fsi_probe(struct platform_device *pdev) master->irq = irq; master->info = pdev->dev.platform_data; master->fsia.base = master->base; + master->fsia.master = master; master->fsib.base = master->base + 0x40; + master->fsib.master = master; pm_runtime_enable(&pdev->dev); pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; + fsi_soc_dai[0].private_data = &master->fsia; fsi_soc_dai[1].dev = &pdev->dev; + fsi_soc_dai[1].private_data = &master->fsib; - fsi_soft_all_reset(); + fsi_soft_all_reset(master); ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); if (ret) { @@ -937,6 +937,10 @@ exit: static int fsi_remove(struct platform_device *pdev) { + struct fsi_master *master; + + master = fsi_get_master(fsi_soc_dai[0].private_data); + snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); @@ -946,7 +950,12 @@ static int fsi_remove(struct platform_device *pdev) iounmap(master->base); kfree(master); - master = NULL; + + fsi_soc_dai[0].dev = NULL; + fsi_soc_dai[0].private_data = NULL; + fsi_soc_dai[1].dev = NULL; + fsi_soc_dai[1].private_data = NULL; + return 0; } -- cgit v0.10.2 From a47979b5aa2117848b742828c98abe7eea42a9ff Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Thu, 3 Dec 2009 18:56:56 +0530 Subject: ASoC: DaVinci: Update suspend/resume support for McASP driver Add clock enable and disable calls to resume and suspend respectively. Also add a member to the audio device data structure which tracks the clock status. Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied. [1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/ 2009-November/016958.html Signed-off-by: Chaithrika U S Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0a302e1..a613bbb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -767,14 +767,27 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, int ret = 0; switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + if (!dev->clk_active) { + clk_enable(dev->clk); + dev->clk_active = 1; + } + /* Fall through */ + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: + davinci_mcasp_stop(dev, substream->stream); + if (dev->clk_active) { + clk_disable(dev->clk); + dev->clk_active = 0; + } + + break; + + case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(dev, substream->stream); break; @@ -866,6 +879,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } clk_enable(dev->clk); + dev->clk_active = 1; dev->base = (void __iomem *)IO_ADDRESS(mem->start); dev->op_mode = pdata->op_mode; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 582c924..e755b51 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -44,6 +44,7 @@ struct davinci_audio_dev { int sample_rate; struct clk *clk; unsigned int codec_fmt; + u8 clk_active; /* McASP specific data */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index ad4d7f4..80c7fdf 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -49,7 +49,7 @@ static void print_buf_info(int slot, char *name) static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE), + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE), .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | -- cgit v0.10.2 From 3a7aaed714bbe3c071000d720f0cce186d1897a4 Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Fri, 4 Dec 2009 13:49:10 +0200 Subject: ASoC: tlv320dac33: Add support for regulator framework Take the regulator framework in use for managing the power sources. Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 9c8903d..5037454 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include #include @@ -58,11 +59,19 @@ enum dac33_state { DAC33_FLUSH, }; +#define DAC33_NUM_SUPPLIES 3 +static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "IOVDD", +}; + struct tlv320dac33_priv { struct mutex mutex; struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec codec; + struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; int power_gpio; int chip_power; int irq; @@ -297,28 +306,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } -static void dac33_hard_power(struct snd_soc_codec *codec, int power) +static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; mutex_lock(&dac33->mutex); if (power) { - if (dac33->power_gpio >= 0) { - gpio_set_value(dac33->power_gpio, 1); - dac33->chip_power = 1; - /* Restore registers */ - dac33_restore_regs(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + if (dac33->power_gpio >= 0) + gpio_set_value(dac33->power_gpio, 1); + + dac33->chip_power = 1; + + /* Restore registers */ + dac33_restore_regs(codec); + dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); - if (dac33->power_gpio >= 0) { + if (dac33->power_gpio >= 0) gpio_set_value(dac33->power_gpio, 0); - dac33->chip_power = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + dac33->chip_power = 0; } - mutex_unlock(&dac33->mutex); +exit: + mutex_unlock(&dac33->mutex); + return ret; } static int dac33_get_nsample(struct snd_kcontrol *kcontrol, @@ -469,6 +499,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + int ret; + switch (level) { case SND_SOC_BIAS_ON: dac33_soft_power(codec, 1); @@ -476,12 +508,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) - dac33_hard_power(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = dac33_hard_power(codec, 1); + if (ret != 0) + return ret; + } + dac33_soft_power(codec, 0); break; case SND_SOC_BIAS_OFF: - dac33_hard_power(codec, 0); + ret = dac33_hard_power(codec, 0); + if (ret != 0) + return ret; + break; } codec->bias_level = level; @@ -959,6 +998,9 @@ static int dac33_soc_probe(struct platform_device *pdev) /* power on device */ dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration has enabled regulator an extra time */ + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); + return 0; pcm_err: @@ -1039,7 +1081,7 @@ static int dac33_i2c_probe(struct i2c_client *client, struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; struct snd_soc_codec *codec; - int ret = 0; + int ret, i; if (client->dev.platform_data == NULL) { dev_err(&client->dev, "Platform data not set\n"); @@ -1130,6 +1172,24 @@ static int dac33_i2c_probe(struct i2c_client *client, } } + for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) + dac33->supplies[i].supply = dac33_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies), + dac33->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), + dac33->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); @@ -1149,6 +1209,10 @@ static int dac33_i2c_probe(struct i2c_client *client, return ret; error_codec: + regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_enable: + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); +err_get: if (dac33->irq >= 0) { free_irq(dac33->irq, &dac33->codec); destroy_workqueue(dac33->dac33_wq); @@ -1177,6 +1241,8 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->irq >= 0) free_irq(dac33->irq, &dac33->codec); + regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); + destroy_workqueue(dac33->dac33_wq); snd_soc_unregister_dai(&dac33_dai); snd_soc_unregister_codec(&dac33->codec); -- cgit v0.10.2 From dd1b3d53c2e5b9cccec9001fc0b63f6b686a4ac9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Dec 2009 14:22:03 +0000 Subject: ASoC: Export snd_soc_update_bits_unlocked() Allows custom controls to use it. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d7718f9..08909cc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -253,6 +253,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, unsigned int value); +int snd_soc_update_bits_locked(struct snd_soc_codec *codec, + unsigned short reg, unsigned int mask, + unsigned int value); int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, unsigned int value); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ef8f282..8b900a84 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1427,9 +1427,9 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * * Returns 1 for change else 0. */ -static int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned short reg, unsigned int mask, - unsigned int value) +int snd_soc_update_bits_locked(struct snd_soc_codec *codec, + unsigned short reg, unsigned int mask, + unsigned int value) { int change; @@ -1439,6 +1439,7 @@ static int snd_soc_update_bits_locked(struct snd_soc_codec *codec, return change; } +EXPORT_SYMBOL_GPL(snd_soc_update_bits_locked); /** * snd_soc_test_bits - test register for change -- cgit v0.10.2 From d033c36ae5cec22c893c710cd026fb732c4086b9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 4 Dec 2009 15:25:56 +0000 Subject: ASoC: Display the power register in DAPM widget debugfs Make it a bit easier to tie DAPM widgets in with the register map without referring to the source by including the register location controlled by the widget. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d294ef..846678a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1147,9 +1147,16 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w); dapm_clear_walk(w->codec); - ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d\n", + ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", w->name, w->power ? "On" : "Off", in, out); + if (w->reg >= 0) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + " - R%d(0x%x) bit %d", + w->reg, w->reg, w->shift); + + ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + if (w->sname) ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, -- cgit v0.10.2 From a91eb199e4dc8a2ab3fb7a53f1a23ce82b29fc04 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Nov 2009 11:56:07 +0000 Subject: ASoC: Initial WM8904 CODEC driver The WM8904 is a high performance ultra-low power stereo CODEC optimised for portable audio applications, with features including a class W amplifier, FLL with free running mode, Mobile ReTune and ground referenced headphone and line outputs. Support for some features, most particularly the digital microphone interface, is not yet present. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h new file mode 100644 index 0000000..d66575a --- /dev/null +++ b/include/sound/wm8904.h @@ -0,0 +1,57 @@ +/* + * Platform data for WM8904 + * + * Copyright 2009 Wolfson Microelectronics PLC. + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __MFD_WM8994_PDATA_H__ +#define __MFD_WM8994_PDATA_H__ + +#define WM8904_DRC_REGS 4 +#define WM8904_EQ_REGS 25 + +/** + * DRC configurations are specified with a label and a set of register + * values to write (the enable bits will be ignored). At runtime an + * enumerated control will be presented for each DRC block allowing + * the user to choose the configration to use. + * + * Configurations may be generated by hand or by using the DRC control + * panel provided by the WISCE - see http://www.wolfsonmicro.com/wisce/ + * for details. + */ +struct wm8904_drc_cfg { + const char *name; + u16 regs[WM8904_DRC_REGS]; +}; + +/** + * ReTune Mobile configurations are specified with a label, sample + * rate and set of values to write (the enable bits will be ignored). + * + * Configurations are expected to be generated using the ReTune Mobile + * control panel in WISCE - see http://www.wolfsonmicro.com/wisce/ + */ +struct wm8904_retune_mobile_cfg { + const char *name; + unsigned int rate; + u16 regs[WM8904_EQ_REGS]; +}; + +struct wm8904_pdata { + int num_drc_cfgs; + struct wm8904_drc_cfg *drc_cfgs; + + int num_retune_mobile_cfgs; + struct wm8904_retune_mobile_cfg *retune_mobile_cfgs; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..011d3ab 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C + select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8960 if I2C select SND_SOC_WM8961 if I2C @@ -203,6 +204,9 @@ config SND_SOC_WM8900 config SND_SOC_WM8903 tristate +config SND_SOC_WM8904 + tristate + config SND_SOC_WM8940 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..0471d90 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -36,6 +36,7 @@ snd-soc-wm8753-objs := wm8753.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o +snd-soc-wm8904-objs := wm8904.o snd-soc-wm8940-objs := wm8940.o snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o @@ -92,6 +93,7 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o +obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c new file mode 100644 index 0000000..8310e5d --- /dev/null +++ b/sound/soc/codecs/wm8904.c @@ -0,0 +1,2538 @@ +/* + * wm8904.c -- WM8904 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8904.h" + +static struct snd_soc_codec *wm8904_codec; +struct snd_soc_codec_device soc_codec_dev_wm8904; + +#define WM8904_NUM_DCS_CHANNELS 4 + +#define WM8904_NUM_SUPPLIES 5 +static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD", + "AVDD", + "CPVDD", + "MICVDD", +}; + +/* codec private data */ +struct wm8904_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8904_MAX_REGISTER + 1]; + + struct regulator_bulk_data supplies[WM8904_NUM_SUPPLIES]; + + struct wm8904_pdata *pdata; + + int deemph; + + /* Platform provided DRC configuration */ + const char **drc_texts; + int drc_cfg; + struct soc_enum drc_enum; + + /* Platform provided ReTune mobile configuration */ + int num_retune_mobile_texts; + const char **retune_mobile_texts; + int retune_mobile_cfg; + struct soc_enum retune_mobile_enum; + + /* FLL setup */ + int fll_src; + int fll_fref; + int fll_fout; + + /* Clocking configuration */ + unsigned int mclk_rate; + int sysclk_src; + unsigned int sysclk_rate; + + int tdm_width; + int tdm_slots; + int bclk; + int fs; + + /* DC servo configuration - cached offset values */ + int dcs_state[WM8904_NUM_DCS_CHANNELS]; +}; + +static const u16 wm8904_reg[WM8904_MAX_REGISTER + 1] = { + 0x8904, /* R0 - SW Reset and ID */ + 0x0000, /* R1 - Revision */ + 0x0000, /* R2 */ + 0x0000, /* R3 */ + 0x0018, /* R4 - Bias Control 0 */ + 0x0000, /* R5 - VMID Control 0 */ + 0x0000, /* R6 - Mic Bias Control 0 */ + 0x0000, /* R7 - Mic Bias Control 1 */ + 0x0001, /* R8 - Analogue DAC 0 */ + 0x9696, /* R9 - mic Filter Control */ + 0x0001, /* R10 - Analogue ADC 0 */ + 0x0000, /* R11 */ + 0x0000, /* R12 - Power Management 0 */ + 0x0000, /* R13 */ + 0x0000, /* R14 - Power Management 2 */ + 0x0000, /* R15 - Power Management 3 */ + 0x0000, /* R16 */ + 0x0000, /* R17 */ + 0x0000, /* R18 - Power Management 6 */ + 0x0000, /* R19 */ + 0x945E, /* R20 - Clock Rates 0 */ + 0x0C05, /* R21 - Clock Rates 1 */ + 0x0006, /* R22 - Clock Rates 2 */ + 0x0000, /* R23 */ + 0x0050, /* R24 - Audio Interface 0 */ + 0x000A, /* R25 - Audio Interface 1 */ + 0x00E4, /* R26 - Audio Interface 2 */ + 0x0040, /* R27 - Audio Interface 3 */ + 0x0000, /* R28 */ + 0x0000, /* R29 */ + 0x00C0, /* R30 - DAC Digital Volume Left */ + 0x00C0, /* R31 - DAC Digital Volume Right */ + 0x0000, /* R32 - DAC Digital 0 */ + 0x0008, /* R33 - DAC Digital 1 */ + 0x0000, /* R34 */ + 0x0000, /* R35 */ + 0x00C0, /* R36 - ADC Digital Volume Left */ + 0x00C0, /* R37 - ADC Digital Volume Right */ + 0x0010, /* R38 - ADC Digital 0 */ + 0x0000, /* R39 - Digital Microphone 0 */ + 0x01AF, /* R40 - DRC 0 */ + 0x3248, /* R41 - DRC 1 */ + 0x0000, /* R42 - DRC 2 */ + 0x0000, /* R43 - DRC 3 */ + 0x0085, /* R44 - Analogue Left Input 0 */ + 0x0085, /* R45 - Analogue Right Input 0 */ + 0x0044, /* R46 - Analogue Left Input 1 */ + 0x0044, /* R47 - Analogue Right Input 1 */ + 0x0000, /* R48 */ + 0x0000, /* R49 */ + 0x0000, /* R50 */ + 0x0000, /* R51 */ + 0x0000, /* R52 */ + 0x0000, /* R53 */ + 0x0000, /* R54 */ + 0x0000, /* R55 */ + 0x0000, /* R56 */ + 0x002D, /* R57 - Analogue OUT1 Left */ + 0x002D, /* R58 - Analogue OUT1 Right */ + 0x0039, /* R59 - Analogue OUT2 Left */ + 0x0039, /* R60 - Analogue OUT2 Right */ + 0x0000, /* R61 - Analogue OUT12 ZC */ + 0x0000, /* R62 */ + 0x0000, /* R63 */ + 0x0000, /* R64 */ + 0x0000, /* R65 */ + 0x0000, /* R66 */ + 0x0000, /* R67 - DC Servo 0 */ + 0x0000, /* R68 - DC Servo 1 */ + 0xAAAA, /* R69 - DC Servo 2 */ + 0x0000, /* R70 */ + 0xAAAA, /* R71 - DC Servo 4 */ + 0xAAAA, /* R72 - DC Servo 5 */ + 0x0000, /* R73 - DC Servo 6 */ + 0x0000, /* R74 - DC Servo 7 */ + 0x0000, /* R75 - DC Servo 8 */ + 0x0000, /* R76 - DC Servo 9 */ + 0x0000, /* R77 - DC Servo Readback 0 */ + 0x0000, /* R78 */ + 0x0000, /* R79 */ + 0x0000, /* R80 */ + 0x0000, /* R81 */ + 0x0000, /* R82 */ + 0x0000, /* R83 */ + 0x0000, /* R84 */ + 0x0000, /* R85 */ + 0x0000, /* R86 */ + 0x0000, /* R87 */ + 0x0000, /* R88 */ + 0x0000, /* R89 */ + 0x0000, /* R90 - Analogue HP 0 */ + 0x0000, /* R91 */ + 0x0000, /* R92 */ + 0x0000, /* R93 */ + 0x0000, /* R94 - Analogue Lineout 0 */ + 0x0000, /* R95 */ + 0x0000, /* R96 */ + 0x0000, /* R97 */ + 0x0000, /* R98 - Charge Pump 0 */ + 0x0000, /* R99 */ + 0x0000, /* R100 */ + 0x0000, /* R101 */ + 0x0000, /* R102 */ + 0x0000, /* R103 */ + 0x0004, /* R104 - Class W 0 */ + 0x0000, /* R105 */ + 0x0000, /* R106 */ + 0x0000, /* R107 */ + 0x0000, /* R108 - Write Sequencer 0 */ + 0x0000, /* R109 - Write Sequencer 1 */ + 0x0000, /* R110 - Write Sequencer 2 */ + 0x0000, /* R111 - Write Sequencer 3 */ + 0x0000, /* R112 - Write Sequencer 4 */ + 0x0000, /* R113 */ + 0x0000, /* R114 */ + 0x0000, /* R115 */ + 0x0000, /* R116 - FLL Control 1 */ + 0x0007, /* R117 - FLL Control 2 */ + 0x0000, /* R118 - FLL Control 3 */ + 0x2EE0, /* R119 - FLL Control 4 */ + 0x0004, /* R120 - FLL Control 5 */ + 0x0014, /* R121 - GPIO Control 1 */ + 0x0010, /* R122 - GPIO Control 2 */ + 0x0010, /* R123 - GPIO Control 3 */ + 0x0000, /* R124 - GPIO Control 4 */ + 0x0000, /* R125 */ + 0x0000, /* R126 - Digital Pulls */ + 0x0000, /* R127 - Interrupt Status */ + 0xFFFF, /* R128 - Interrupt Status Mask */ + 0x0000, /* R129 - Interrupt Polarity */ + 0x0000, /* R130 - Interrupt Debounce */ + 0x0000, /* R131 */ + 0x0000, /* R132 */ + 0x0000, /* R133 */ + 0x0000, /* R134 - EQ1 */ + 0x000C, /* R135 - EQ2 */ + 0x000C, /* R136 - EQ3 */ + 0x000C, /* R137 - EQ4 */ + 0x000C, /* R138 - EQ5 */ + 0x000C, /* R139 - EQ6 */ + 0x0FCA, /* R140 - EQ7 */ + 0x0400, /* R141 - EQ8 */ + 0x00D8, /* R142 - EQ9 */ + 0x1EB5, /* R143 - EQ10 */ + 0xF145, /* R144 - EQ11 */ + 0x0B75, /* R145 - EQ12 */ + 0x01C5, /* R146 - EQ13 */ + 0x1C58, /* R147 - EQ14 */ + 0xF373, /* R148 - EQ15 */ + 0x0A54, /* R149 - EQ16 */ + 0x0558, /* R150 - EQ17 */ + 0x168E, /* R151 - EQ18 */ + 0xF829, /* R152 - EQ19 */ + 0x07AD, /* R153 - EQ20 */ + 0x1103, /* R154 - EQ21 */ + 0x0564, /* R155 - EQ22 */ + 0x0559, /* R156 - EQ23 */ + 0x4000, /* R157 - EQ24 */ + 0x0000, /* R158 */ + 0x0000, /* R159 */ + 0x0000, /* R160 */ + 0x0000, /* R161 - Control Interface Test 1 */ + 0x0000, /* R162 */ + 0x0000, /* R163 */ + 0x0000, /* R164 */ + 0x0000, /* R165 */ + 0x0000, /* R166 */ + 0x0000, /* R167 */ + 0x0000, /* R168 */ + 0x0000, /* R169 */ + 0x0000, /* R170 */ + 0x0000, /* R171 */ + 0x0000, /* R172 */ + 0x0000, /* R173 */ + 0x0000, /* R174 */ + 0x0000, /* R175 */ + 0x0000, /* R176 */ + 0x0000, /* R177 */ + 0x0000, /* R178 */ + 0x0000, /* R179 */ + 0x0000, /* R180 */ + 0x0000, /* R181 */ + 0x0000, /* R182 */ + 0x0000, /* R183 */ + 0x0000, /* R184 */ + 0x0000, /* R185 */ + 0x0000, /* R186 */ + 0x0000, /* R187 */ + 0x0000, /* R188 */ + 0x0000, /* R189 */ + 0x0000, /* R190 */ + 0x0000, /* R191 */ + 0x0000, /* R192 */ + 0x0000, /* R193 */ + 0x0000, /* R194 */ + 0x0000, /* R195 */ + 0x0000, /* R196 */ + 0x0000, /* R197 */ + 0x0000, /* R198 */ + 0x0000, /* R199 */ + 0x0000, /* R200 */ + 0x0000, /* R201 */ + 0x0000, /* R202 */ + 0x0000, /* R203 */ + 0x0000, /* R204 - Analogue Output Bias 0 */ + 0x0000, /* R205 */ + 0x0000, /* R206 */ + 0x0000, /* R207 */ + 0x0000, /* R208 */ + 0x0000, /* R209 */ + 0x0000, /* R210 */ + 0x0000, /* R211 */ + 0x0000, /* R212 */ + 0x0000, /* R213 */ + 0x0000, /* R214 */ + 0x0000, /* R215 */ + 0x0000, /* R216 */ + 0x0000, /* R217 */ + 0x0000, /* R218 */ + 0x0000, /* R219 */ + 0x0000, /* R220 */ + 0x0000, /* R221 */ + 0x0000, /* R222 */ + 0x0000, /* R223 */ + 0x0000, /* R224 */ + 0x0000, /* R225 */ + 0x0000, /* R226 */ + 0x0000, /* R227 */ + 0x0000, /* R228 */ + 0x0000, /* R229 */ + 0x0000, /* R230 */ + 0x0000, /* R231 */ + 0x0000, /* R232 */ + 0x0000, /* R233 */ + 0x0000, /* R234 */ + 0x0000, /* R235 */ + 0x0000, /* R236 */ + 0x0000, /* R237 */ + 0x0000, /* R238 */ + 0x0000, /* R239 */ + 0x0000, /* R240 */ + 0x0000, /* R241 */ + 0x0000, /* R242 */ + 0x0000, /* R243 */ + 0x0000, /* R244 */ + 0x0000, /* R245 */ + 0x0000, /* R246 */ + 0x0000, /* R247 - FLL NCO Test 0 */ + 0x0019, /* R248 - FLL NCO Test 1 */ +}; + +static struct { + int readable; + int writable; + int vol; +} wm8904_access[] = { + { 0xFFFF, 0xFFFF, 1 }, /* R0 - SW Reset and ID */ + { 0x0000, 0x0000, 0 }, /* R1 - Revision */ + { 0x0000, 0x0000, 0 }, /* R2 */ + { 0x0000, 0x0000, 0 }, /* R3 */ + { 0x001F, 0x001F, 0 }, /* R4 - Bias Control 0 */ + { 0x0047, 0x0047, 0 }, /* R5 - VMID Control 0 */ + { 0x007F, 0x007F, 0 }, /* R6 - Mic Bias Control 0 */ + { 0xC007, 0xC007, 0 }, /* R7 - Mic Bias Control 1 */ + { 0x001E, 0x001E, 0 }, /* R8 - Analogue DAC 0 */ + { 0xFFFF, 0xFFFF, 0 }, /* R9 - mic Filter Control */ + { 0x0001, 0x0001, 0 }, /* R10 - Analogue ADC 0 */ + { 0x0000, 0x0000, 0 }, /* R11 */ + { 0x0003, 0x0003, 0 }, /* R12 - Power Management 0 */ + { 0x0000, 0x0000, 0 }, /* R13 */ + { 0x0003, 0x0003, 0 }, /* R14 - Power Management 2 */ + { 0x0003, 0x0003, 0 }, /* R15 - Power Management 3 */ + { 0x0000, 0x0000, 0 }, /* R16 */ + { 0x0000, 0x0000, 0 }, /* R17 */ + { 0x000F, 0x000F, 0 }, /* R18 - Power Management 6 */ + { 0x0000, 0x0000, 0 }, /* R19 */ + { 0x7001, 0x7001, 0 }, /* R20 - Clock Rates 0 */ + { 0x3C07, 0x3C07, 0 }, /* R21 - Clock Rates 1 */ + { 0xD00F, 0xD00F, 0 }, /* R22 - Clock Rates 2 */ + { 0x0000, 0x0000, 0 }, /* R23 */ + { 0x1FFF, 0x1FFF, 0 }, /* R24 - Audio Interface 0 */ + { 0x3DDF, 0x3DDF, 0 }, /* R25 - Audio Interface 1 */ + { 0x0F1F, 0x0F1F, 0 }, /* R26 - Audio Interface 2 */ + { 0x0FFF, 0x0FFF, 0 }, /* R27 - Audio Interface 3 */ + { 0x0000, 0x0000, 0 }, /* R28 */ + { 0x0000, 0x0000, 0 }, /* R29 */ + { 0x00FF, 0x01FF, 0 }, /* R30 - DAC Digital Volume Left */ + { 0x00FF, 0x01FF, 0 }, /* R31 - DAC Digital Volume Right */ + { 0x0FFF, 0x0FFF, 0 }, /* R32 - DAC Digital 0 */ + { 0x1E4E, 0x1E4E, 0 }, /* R33 - DAC Digital 1 */ + { 0x0000, 0x0000, 0 }, /* R34 */ + { 0x0000, 0x0000, 0 }, /* R35 */ + { 0x00FF, 0x01FF, 0 }, /* R36 - ADC Digital Volume Left */ + { 0x00FF, 0x01FF, 0 }, /* R37 - ADC Digital Volume Right */ + { 0x0073, 0x0073, 0 }, /* R38 - ADC Digital 0 */ + { 0x1800, 0x1800, 0 }, /* R39 - Digital Microphone 0 */ + { 0xDFEF, 0xDFEF, 0 }, /* R40 - DRC 0 */ + { 0xFFFF, 0xFFFF, 0 }, /* R41 - DRC 1 */ + { 0x003F, 0x003F, 0 }, /* R42 - DRC 2 */ + { 0x07FF, 0x07FF, 0 }, /* R43 - DRC 3 */ + { 0x009F, 0x009F, 0 }, /* R44 - Analogue Left Input 0 */ + { 0x009F, 0x009F, 0 }, /* R45 - Analogue Right Input 0 */ + { 0x007F, 0x007F, 0 }, /* R46 - Analogue Left Input 1 */ + { 0x007F, 0x007F, 0 }, /* R47 - Analogue Right Input 1 */ + { 0x0000, 0x0000, 0 }, /* R48 */ + { 0x0000, 0x0000, 0 }, /* R49 */ + { 0x0000, 0x0000, 0 }, /* R50 */ + { 0x0000, 0x0000, 0 }, /* R51 */ + { 0x0000, 0x0000, 0 }, /* R52 */ + { 0x0000, 0x0000, 0 }, /* R53 */ + { 0x0000, 0x0000, 0 }, /* R54 */ + { 0x0000, 0x0000, 0 }, /* R55 */ + { 0x0000, 0x0000, 0 }, /* R56 */ + { 0x017F, 0x01FF, 0 }, /* R57 - Analogue OUT1 Left */ + { 0x017F, 0x01FF, 0 }, /* R58 - Analogue OUT1 Right */ + { 0x017F, 0x01FF, 0 }, /* R59 - Analogue OUT2 Left */ + { 0x017F, 0x01FF, 0 }, /* R60 - Analogue OUT2 Right */ + { 0x000F, 0x000F, 0 }, /* R61 - Analogue OUT12 ZC */ + { 0x0000, 0x0000, 0 }, /* R62 */ + { 0x0000, 0x0000, 0 }, /* R63 */ + { 0x0000, 0x0000, 0 }, /* R64 */ + { 0x0000, 0x0000, 0 }, /* R65 */ + { 0x0000, 0x0000, 0 }, /* R66 */ + { 0x000F, 0x000F, 0 }, /* R67 - DC Servo 0 */ + { 0xFFFF, 0xFFFF, 1 }, /* R68 - DC Servo 1 */ + { 0x0F0F, 0x0F0F, 0 }, /* R69 - DC Servo 2 */ + { 0x0000, 0x0000, 0 }, /* R70 */ + { 0x007F, 0x007F, 0 }, /* R71 - DC Servo 4 */ + { 0x007F, 0x007F, 0 }, /* R72 - DC Servo 5 */ + { 0x00FF, 0x00FF, 1 }, /* R73 - DC Servo 6 */ + { 0x00FF, 0x00FF, 1 }, /* R74 - DC Servo 7 */ + { 0x00FF, 0x00FF, 1 }, /* R75 - DC Servo 8 */ + { 0x00FF, 0x00FF, 1 }, /* R76 - DC Servo 9 */ + { 0x0FFF, 0x0000, 1 }, /* R77 - DC Servo Readback 0 */ + { 0x0000, 0x0000, 0 }, /* R78 */ + { 0x0000, 0x0000, 0 }, /* R79 */ + { 0x0000, 0x0000, 0 }, /* R80 */ + { 0x0000, 0x0000, 0 }, /* R81 */ + { 0x0000, 0x0000, 0 }, /* R82 */ + { 0x0000, 0x0000, 0 }, /* R83 */ + { 0x0000, 0x0000, 0 }, /* R84 */ + { 0x0000, 0x0000, 0 }, /* R85 */ + { 0x0000, 0x0000, 0 }, /* R86 */ + { 0x0000, 0x0000, 0 }, /* R87 */ + { 0x0000, 0x0000, 0 }, /* R88 */ + { 0x0000, 0x0000, 0 }, /* R89 */ + { 0x00FF, 0x00FF, 0 }, /* R90 - Analogue HP 0 */ + { 0x0000, 0x0000, 0 }, /* R91 */ + { 0x0000, 0x0000, 0 }, /* R92 */ + { 0x0000, 0x0000, 0 }, /* R93 */ + { 0x00FF, 0x00FF, 0 }, /* R94 - Analogue Lineout 0 */ + { 0x0000, 0x0000, 0 }, /* R95 */ + { 0x0000, 0x0000, 0 }, /* R96 */ + { 0x0000, 0x0000, 0 }, /* R97 */ + { 0x0001, 0x0001, 0 }, /* R98 - Charge Pump 0 */ + { 0x0000, 0x0000, 0 }, /* R99 */ + { 0x0000, 0x0000, 0 }, /* R100 */ + { 0x0000, 0x0000, 0 }, /* R101 */ + { 0x0000, 0x0000, 0 }, /* R102 */ + { 0x0000, 0x0000, 0 }, /* R103 */ + { 0x0001, 0x0001, 0 }, /* R104 - Class W 0 */ + { 0x0000, 0x0000, 0 }, /* R105 */ + { 0x0000, 0x0000, 0 }, /* R106 */ + { 0x0000, 0x0000, 0 }, /* R107 */ + { 0x011F, 0x011F, 0 }, /* R108 - Write Sequencer 0 */ + { 0x7FFF, 0x7FFF, 0 }, /* R109 - Write Sequencer 1 */ + { 0x4FFF, 0x4FFF, 0 }, /* R110 - Write Sequencer 2 */ + { 0x003F, 0x033F, 0 }, /* R111 - Write Sequencer 3 */ + { 0x03F1, 0x0000, 0 }, /* R112 - Write Sequencer 4 */ + { 0x0000, 0x0000, 0 }, /* R113 */ + { 0x0000, 0x0000, 0 }, /* R114 */ + { 0x0000, 0x0000, 0 }, /* R115 */ + { 0x0007, 0x0007, 0 }, /* R116 - FLL Control 1 */ + { 0x3F77, 0x3F77, 0 }, /* R117 - FLL Control 2 */ + { 0xFFFF, 0xFFFF, 0 }, /* R118 - FLL Control 3 */ + { 0x7FEF, 0x7FEF, 0 }, /* R119 - FLL Control 4 */ + { 0x001B, 0x001B, 0 }, /* R120 - FLL Control 5 */ + { 0x003F, 0x003F, 0 }, /* R121 - GPIO Control 1 */ + { 0x003F, 0x003F, 0 }, /* R122 - GPIO Control 2 */ + { 0x003F, 0x003F, 0 }, /* R123 - GPIO Control 3 */ + { 0x038F, 0x038F, 0 }, /* R124 - GPIO Control 4 */ + { 0x0000, 0x0000, 0 }, /* R125 */ + { 0x00FF, 0x00FF, 0 }, /* R126 - Digital Pulls */ + { 0x07FF, 0x03FF, 1 }, /* R127 - Interrupt Status */ + { 0x03FF, 0x03FF, 0 }, /* R128 - Interrupt Status Mask */ + { 0x03FF, 0x03FF, 0 }, /* R129 - Interrupt Polarity */ + { 0x03FF, 0x03FF, 0 }, /* R130 - Interrupt Debounce */ + { 0x0000, 0x0000, 0 }, /* R131 */ + { 0x0000, 0x0000, 0 }, /* R132 */ + { 0x0000, 0x0000, 0 }, /* R133 */ + { 0x0001, 0x0001, 0 }, /* R134 - EQ1 */ + { 0x001F, 0x001F, 0 }, /* R135 - EQ2 */ + { 0x001F, 0x001F, 0 }, /* R136 - EQ3 */ + { 0x001F, 0x001F, 0 }, /* R137 - EQ4 */ + { 0x001F, 0x001F, 0 }, /* R138 - EQ5 */ + { 0x001F, 0x001F, 0 }, /* R139 - EQ6 */ + { 0xFFFF, 0xFFFF, 0 }, /* R140 - EQ7 */ + { 0xFFFF, 0xFFFF, 0 }, /* R141 - EQ8 */ + { 0xFFFF, 0xFFFF, 0 }, /* R142 - EQ9 */ + { 0xFFFF, 0xFFFF, 0 }, /* R143 - EQ10 */ + { 0xFFFF, 0xFFFF, 0 }, /* R144 - EQ11 */ + { 0xFFFF, 0xFFFF, 0 }, /* R145 - EQ12 */ + { 0xFFFF, 0xFFFF, 0 }, /* R146 - EQ13 */ + { 0xFFFF, 0xFFFF, 0 }, /* R147 - EQ14 */ + { 0xFFFF, 0xFFFF, 0 }, /* R148 - EQ15 */ + { 0xFFFF, 0xFFFF, 0 }, /* R149 - EQ16 */ + { 0xFFFF, 0xFFFF, 0 }, /* R150 - EQ17 */ + { 0xFFFF, 0xFFFF, 0 }, /* R151wm8523_dai - EQ18 */ + { 0xFFFF, 0xFFFF, 0 }, /* R152 - EQ19 */ + { 0xFFFF, 0xFFFF, 0 }, /* R153 - EQ20 */ + { 0xFFFF, 0xFFFF, 0 }, /* R154 - EQ21 */ + { 0xFFFF, 0xFFFF, 0 }, /* R155 - EQ22 */ + { 0xFFFF, 0xFFFF, 0 }, /* R156 - EQ23 */ + { 0xFFFF, 0xFFFF, 0 }, /* R157 - EQ24 */ + { 0x0000, 0x0000, 0 }, /* R158 */ + { 0x0000, 0x0000, 0 }, /* R159 */ + { 0x0000, 0x0000, 0 }, /* R160 */ + { 0x0002, 0x0002, 0 }, /* R161 - Control Interface Test 1 */ + { 0x0000, 0x0000, 0 }, /* R162 */ + { 0x0000, 0x0000, 0 }, /* R163 */ + { 0x0000, 0x0000, 0 }, /* R164 */ + { 0x0000, 0x0000, 0 }, /* R165 */ + { 0x0000, 0x0000, 0 }, /* R166 */ + { 0x0000, 0x0000, 0 }, /* R167 */ + { 0x0000, 0x0000, 0 }, /* R168 */ + { 0x0000, 0x0000, 0 }, /* R169 */ + { 0x0000, 0x0000, 0 }, /* R170 */ + { 0x0000, 0x0000, 0 }, /* R171 */ + { 0x0000, 0x0000, 0 }, /* R172 */ + { 0x0000, 0x0000, 0 }, /* R173 */ + { 0x0000, 0x0000, 0 }, /* R174 */ + { 0x0000, 0x0000, 0 }, /* R175 */ + { 0x0000, 0x0000, 0 }, /* R176 */ + { 0x0000, 0x0000, 0 }, /* R177 */ + { 0x0000, 0x0000, 0 }, /* R178 */ + { 0x0000, 0x0000, 0 }, /* R179 */ + { 0x0000, 0x0000, 0 }, /* R180 */ + { 0x0000, 0x0000, 0 }, /* R181 */ + { 0x0000, 0x0000, 0 }, /* R182 */ + { 0x0000, 0x0000, 0 }, /* R183 */ + { 0x0000, 0x0000, 0 }, /* R184 */ + { 0x0000, 0x0000, 0 }, /* R185 */ + { 0x0000, 0x0000, 0 }, /* R186 */ + { 0x0000, 0x0000, 0 }, /* R187 */ + { 0x0000, 0x0000, 0 }, /* R188 */ + { 0x0000, 0x0000, 0 }, /* R189 */ + { 0x0000, 0x0000, 0 }, /* R190 */ + { 0x0000, 0x0000, 0 }, /* R191 */ + { 0x0000, 0x0000, 0 }, /* R192 */ + { 0x0000, 0x0000, 0 }, /* R193 */ + { 0x0000, 0x0000, 0 }, /* R194 */ + { 0x0000, 0x0000, 0 }, /* R195 */ + { 0x0000, 0x0000, 0 }, /* R196 */ + { 0x0000, 0x0000, 0 }, /* R197 */ + { 0x0000, 0x0000, 0 }, /* R198 */ + { 0x0000, 0x0000, 0 }, /* R199 */ + { 0x0000, 0x0000, 0 }, /* R200 */ + { 0x0000, 0x0000, 0 }, /* R201 */ + { 0x0000, 0x0000, 0 }, /* R202 */ + { 0x0000, 0x0000, 0 }, /* R203 */ + { 0x0070, 0x0070, 0 }, /* R204 - Analogue Output Bias 0 */ + { 0x0000, 0x0000, 0 }, /* R205 */ + { 0x0000, 0x0000, 0 }, /* R206 */ + { 0x0000, 0x0000, 0 }, /* R207 */ + { 0x0000, 0x0000, 0 }, /* R208 */ + { 0x0000, 0x0000, 0 }, /* R209 */ + { 0x0000, 0x0000, 0 }, /* R210 */ + { 0x0000, 0x0000, 0 }, /* R211 */ + { 0x0000, 0x0000, 0 }, /* R212 */ + { 0x0000, 0x0000, 0 }, /* R213 */ + { 0x0000, 0x0000, 0 }, /* R214 */ + { 0x0000, 0x0000, 0 }, /* R215 */ + { 0x0000, 0x0000, 0 }, /* R216 */ + { 0x0000, 0x0000, 0 }, /* R217 */ + { 0x0000, 0x0000, 0 }, /* R218 */ + { 0x0000, 0x0000, 0 }, /* R219 */ + { 0x0000, 0x0000, 0 }, /* R220 */ + { 0x0000, 0x0000, 0 }, /* R221 */ + { 0x0000, 0x0000, 0 }, /* R222 */ + { 0x0000, 0x0000, 0 }, /* R223 */ + { 0x0000, 0x0000, 0 }, /* R224 */ + { 0x0000, 0x0000, 0 }, /* R225 */ + { 0x0000, 0x0000, 0 }, /* R226 */ + { 0x0000, 0x0000, 0 }, /* R227 */ + { 0x0000, 0x0000, 0 }, /* R228 */ + { 0x0000, 0x0000, 0 }, /* R229 */ + { 0x0000, 0x0000, 0 }, /* R230 */ + { 0x0000, 0x0000, 0 }, /* R231 */ + { 0x0000, 0x0000, 0 }, /* R232 */ + { 0x0000, 0x0000, 0 }, /* R233 */ + { 0x0000, 0x0000, 0 }, /* R234 */ + { 0x0000, 0x0000, 0 }, /* R235 */ + { 0x0000, 0x0000, 0 }, /* R236 */ + { 0x0000, 0x0000, 0 }, /* R237 */ + { 0x0000, 0x0000, 0 }, /* R238 */ + { 0x0000, 0x0000, 0 }, /* R239 */ + { 0x0000, 0x0000, 0 }, /* R240 */ + { 0x0000, 0x0000, 0 }, /* R241 */ + { 0x0000, 0x0000, 0 }, /* R242 */ + { 0x0000, 0x0000, 0 }, /* R243 */ + { 0x0000, 0x0000, 0 }, /* R244 */ + { 0x0000, 0x0000, 0 }, /* R245 */ + { 0x0000, 0x0000, 0 }, /* R246 */ + { 0x0001, 0x0001, 0 }, /* R247 - FLL NCO Test 0 */ + { 0x003F, 0x003F, 0 }, /* R248 - FLL NCO Test 1 */ +}; + +static int wm8904_volatile_register(unsigned int reg) +{ + return wm8904_access[reg].vol; +} + +static int wm8904_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8904_SW_RESET_AND_ID, 0); +} + +static int wm8904_configure_clocking(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + unsigned int clock0, clock2, rate; + + /* Gate the clock while we're updating to avoid misclocking */ + clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_SYSCLK_SRC, 0); + + /* This should be done on init() for bypass paths */ + switch (wm8904->sysclk_src) { + case WM8904_CLK_MCLK: + dev_dbg(codec->dev, "Using %dHz MCLK\n", wm8904->mclk_rate); + + clock2 &= ~WM8904_SYSCLK_SRC; + rate = wm8904->mclk_rate; + + /* Ensure the FLL is stopped */ + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0); + break; + + case WM8904_CLK_FLL: + dev_dbg(codec->dev, "Using %dHz FLL clock\n", + wm8904->fll_fout); + + clock2 |= WM8904_SYSCLK_SRC; + rate = wm8904->fll_fout; + break; + + default: + dev_err(codec->dev, "System clock not configured\n"); + return -EINVAL; + } + + /* SYSCLK shouldn't be over 13.5MHz */ + if (rate > 13500000) { + clock0 = WM8904_MCLK_DIV; + wm8904->sysclk_rate = rate / 2; + } else { + clock0 = 0; + wm8904->sysclk_rate = rate; + } + + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, WM8904_MCLK_DIV, + clock0); + + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_CLK_SYS_ENA | WM8904_SYSCLK_SRC, clock2); + + dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm8904->sysclk_rate); + + return 0; +} + +static void wm8904_set_drc(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + struct wm8904_pdata *pdata = wm8904->pdata; + int save, i; + + /* Save any enables; the configuration should clear them. */ + save = snd_soc_read(codec, WM8904_DRC_0); + + for (i = 0; i < WM8904_DRC_REGS; i++) + snd_soc_update_bits(codec, WM8904_DRC_0 + i, 0xffff, + pdata->drc_cfgs[wm8904->drc_cfg].regs[i]); + + /* Reenable the DRC */ + snd_soc_update_bits(codec, WM8904_DRC_0, + WM8904_DRC_ENA | WM8904_DRC_DAC_PATH, save); +} + +static int wm8904_put_drc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + struct wm8904_pdata *pdata = wm8904->pdata; + int value = ucontrol->value.integer.value[0]; + + if (value >= pdata->num_drc_cfgs) + return -EINVAL; + + wm8904->drc_cfg = value; + + wm8904_set_drc(codec); + + return 0; +} + +static int wm8904_get_drc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + + ucontrol->value.enumerated.item[0] = wm8904->drc_cfg; + + return 0; +} + +static void wm8904_set_retune_mobile(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + struct wm8904_pdata *pdata = wm8904->pdata; + int best, best_val, save, i, cfg; + + if (!pdata || !wm8904->num_retune_mobile_texts) + return; + + /* Find the version of the currently selected configuration + * with the nearest sample rate. */ + cfg = wm8904->retune_mobile_cfg; + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8904->retune_mobile_texts[cfg]) == 0 && + abs(pdata->retune_mobile_cfgs[i].rate + - wm8904->fs) < best_val) { + best = i; + best_val = abs(pdata->retune_mobile_cfgs[i].rate + - wm8904->fs); + } + } + + dev_dbg(codec->dev, "ReTune Mobile %s/%dHz for %dHz sample rate\n", + pdata->retune_mobile_cfgs[best].name, + pdata->retune_mobile_cfgs[best].rate, + wm8904->fs); + + /* The EQ will be disabled while reconfiguring it, remember the + * current configuration. + */ + save = snd_soc_read(codec, WM8904_EQ1); + + for (i = 0; i < WM8904_EQ_REGS; i++) + snd_soc_update_bits(codec, WM8904_EQ1 + i, 0xffff, + pdata->retune_mobile_cfgs[best].regs[i]); + + snd_soc_update_bits(codec, WM8904_EQ1, WM8904_EQ_ENA, save); +} + +static int wm8904_put_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + struct wm8904_pdata *pdata = wm8904->pdata; + int value = ucontrol->value.integer.value[0]; + + if (value >= pdata->num_retune_mobile_cfgs) + return -EINVAL; + + wm8904->retune_mobile_cfg = value; + + wm8904_set_retune_mobile(codec); + + return 0; +} + +static int wm8904_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + + ucontrol->value.enumerated.item[0] = wm8904->retune_mobile_cfg; + + return 0; +} + +static int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int wm8904_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + int val, i, best; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8904->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - wm8904->fs) < + abs(deemph_settings[best] - wm8904->fs)) + best = i; + } + + val = best << WM8904_DEEMPH_SHIFT; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1, + WM8904_DEEMPH_MASK, val); +} + +static int wm8904_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + + return wm8904->deemph; +} + +static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8904_priv *wm8904 = codec->private_data; + int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + wm8904->deemph = deemph; + + return wm8904_set_deemph(codec); +} + +static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); + +static const char *input_mode_text[] = { + "Single-Ended", "Differential Line", "Differential Mic" +}; + +static const struct soc_enum lin_mode = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text); + +static const struct soc_enum rin_mode = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text); + +static const char *hpf_mode_text[] = { + "Hi-fi", "Voice 1", "Voice 2", "Voice 3" +}; + +static const struct soc_enum hpf_mode = + SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text); + +static const struct snd_kcontrol_new wm8904_adc_snd_controls[] = { +SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_DIGITAL_VOLUME_RIGHT, 1, 119, 0, digital_tlv), + +SOC_ENUM("Left Caputure Mode", lin_mode), +SOC_ENUM("Right Capture Mode", rin_mode), + +/* No TLV since it depends on mode */ +SOC_DOUBLE_R("Capture Volume", WM8904_ANALOGUE_LEFT_INPUT_0, + WM8904_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0, + WM8904_ANALOGUE_RIGHT_INPUT_0, 7, 1, 0), + +SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0), +SOC_ENUM("High Pass Filter Mode", hpf_mode), + +SOC_SINGLE("ADC 128x OSR Switch", WM8904_ANALOGUE_ADC_0, 0, 1, 0), +}; + +static const char *drc_path_text[] = { + "ADC", "DAC" +}; + +static const struct soc_enum drc_path = + SOC_ENUM_SINGLE(WM8904_DRC_0, 14, 2, drc_path_text); + +static const struct snd_kcontrol_new wm8904_dac_snd_controls[] = { +SOC_SINGLE_TLV("Digital Playback Boost Volume", + WM8904_AUDIO_INTERFACE_0, 9, 3, 0, dac_boost_tlv), +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv), + +SOC_DOUBLE_R_TLV("Headphone Volume", WM8904_ANALOGUE_OUT1_LEFT, + WM8904_ANALOGUE_OUT1_RIGHT, 0, 63, 0, out_tlv), +SOC_DOUBLE_R("Headphone Switch", WM8904_ANALOGUE_OUT1_LEFT, + WM8904_ANALOGUE_OUT1_RIGHT, 8, 1, 1), +SOC_DOUBLE_R("Headphone ZC Switch", WM8904_ANALOGUE_OUT1_LEFT, + WM8904_ANALOGUE_OUT1_RIGHT, 6, 1, 0), + +SOC_DOUBLE_R_TLV("Line Output Volume", WM8904_ANALOGUE_OUT2_LEFT, + WM8904_ANALOGUE_OUT2_RIGHT, 0, 63, 0, out_tlv), +SOC_DOUBLE_R("Line Output Switch", WM8904_ANALOGUE_OUT2_LEFT, + WM8904_ANALOGUE_OUT2_RIGHT, 8, 1, 1), +SOC_DOUBLE_R("Line Output ZC Switch", WM8904_ANALOGUE_OUT2_LEFT, + WM8904_ANALOGUE_OUT2_RIGHT, 6, 1, 0), + +SOC_SINGLE("EQ Switch", WM8904_EQ1, 0, 1, 0), +SOC_SINGLE("DRC Switch", WM8904_DRC_0, 15, 1, 0), +SOC_ENUM("DRC Path", drc_path), +SOC_SINGLE("DAC OSRx2 Switch", WM8904_DAC_DIGITAL_1, 6, 1, 0), +SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + wm8904_get_deemph, wm8904_put_deemph), +}; + +static const struct snd_kcontrol_new wm8904_snd_controls[] = { +SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8904_DAC_DIGITAL_0, 4, 8, 15, 0, + sidetone_tlv), +}; + +static const struct snd_kcontrol_new wm8904_eq_controls[] = { +SOC_SINGLE_TLV("EQ1 Volume", WM8904_EQ2, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Volume", WM8904_EQ3, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Volume", WM8904_EQ4, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Volume", WM8904_EQ5, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ5 Volume", WM8904_EQ6, 0, 24, 0, eq_tlv), +}; + +static int cp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + BUG_ON(event != SND_SOC_DAPM_POST_PMU); + + /* Maximum startup time */ + udelay(500); + + return 0; +} + +static int sysclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8904_priv *wm8904 = codec->private_data; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* If we're using the FLL then we only start it when + * required; we assume that the configuration has been + * done previously and all we need to do is kick it + * off. + */ + switch (wm8904->sysclk_src) { + case WM8904_CLK_FLL: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA, + WM8904_FLL_OSC_ENA); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_ENA, + WM8904_FLL_ENA); + break; + + default: + break; + } + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0); + break; + } + + return 0; +} + +static int out_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8904_priv *wm8904 = codec->private_data; + int reg, val; + int dcs_mask; + int dcs_l, dcs_r; + int dcs_l_reg, dcs_r_reg; + int timeout; + + /* This code is shared between HP and LINEOUT; we do all our + * power management in stereo pairs to avoid latency issues so + * we reuse shift to identify which rather than strcmp() the + * name. */ + reg = w->shift; + + switch (reg) { + case WM8904_ANALOGUE_HP_0: + dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1; + dcs_r_reg = WM8904_DC_SERVO_8; + dcs_l_reg = WM8904_DC_SERVO_9; + dcs_l = 0; + dcs_r = 1; + break; + case WM8904_ANALOGUE_LINEOUT_0: + dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3; + dcs_r_reg = WM8904_DC_SERVO_6; + dcs_l_reg = WM8904_DC_SERVO_7; + dcs_l = 2; + dcs_r = 3; + break; + default: + BUG(); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Power on the amplifier */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_ENA | WM8904_HPR_ENA, + WM8904_HPL_ENA | WM8904_HPR_ENA); + + /* Enable the first stage */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY, + WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY); + + /* Power up the DC servo */ + snd_soc_update_bits(codec, WM8904_DC_SERVO_0, + dcs_mask, dcs_mask); + + /* Either calibrate the DC servo or restore cached state + * if we have that. + */ + if (wm8904->dcs_state[dcs_l] || wm8904->dcs_state[dcs_r]) { + dev_dbg(codec->dev, "Restoring DC servo state\n"); + + snd_soc_write(codec, dcs_l_reg, + wm8904->dcs_state[dcs_l]); + snd_soc_write(codec, dcs_r_reg, + wm8904->dcs_state[dcs_r]); + + snd_soc_write(codec, WM8904_DC_SERVO_1, dcs_mask); + + timeout = 20; + } else { + dev_dbg(codec->dev, "Calibrating DC servo\n"); + + snd_soc_write(codec, WM8904_DC_SERVO_1, + dcs_mask << WM8904_DCS_TRIG_STARTUP_0_SHIFT); + + timeout = 500; + } + + /* Wait for DC servo to complete */ + dcs_mask <<= WM8904_DCS_CAL_COMPLETE_SHIFT; + do { + val = snd_soc_read(codec, WM8904_DC_SERVO_READBACK_0); + if ((val & dcs_mask) == dcs_mask) + break; + + msleep(1); + } while (--timeout); + + if ((val & dcs_mask) != dcs_mask) + dev_warn(codec->dev, "DC servo timed out\n"); + else + dev_dbg(codec->dev, "DC servo ready\n"); + + /* Enable the output stage */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, + WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP); + + /* Unshort the output itself */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_RMV_SHORT | + WM8904_HPR_RMV_SHORT, + WM8904_HPL_RMV_SHORT | + WM8904_HPR_RMV_SHORT); + + break; + + case SND_SOC_DAPM_PRE_PMD: + /* Short the output */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_RMV_SHORT | + WM8904_HPR_RMV_SHORT, 0); + + /* Cache the DC servo configuration; this will be + * invalidated if we change the configuration. */ + wm8904->dcs_state[dcs_l] = snd_soc_read(codec, dcs_l_reg); + wm8904->dcs_state[dcs_r] = snd_soc_read(codec, dcs_r_reg); + + snd_soc_update_bits(codec, WM8904_DC_SERVO_0, + dcs_mask, 0); + + /* Disable the amplifier input and output stages */ + snd_soc_update_bits(codec, reg, + WM8904_HPL_ENA | WM8904_HPR_ENA | + WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY | + WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, + 0); + break; + } + + return 0; +} + +static const char *lin_text[] = { + "IN1L", "IN2L", "IN3L" +}; + +static const struct soc_enum lin_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 2, 3, lin_text); + +static const struct snd_kcontrol_new lin_mux = + SOC_DAPM_ENUM("Left Capture Mux", lin_enum); + +static const struct soc_enum lin_inv_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 4, 3, lin_text); + +static const struct snd_kcontrol_new lin_inv_mux = + SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); + +static const char *rin_text[] = { + "IN1R", "IN2R", "IN3R" +}; + +static const struct soc_enum rin_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 2, 3, rin_text); + +static const struct snd_kcontrol_new rin_mux = + SOC_DAPM_ENUM("Right Capture Mux", rin_enum); + +static const struct soc_enum rin_inv_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 4, 3, rin_text); + +static const struct snd_kcontrol_new rin_inv_mux = + SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); + +static const char *aif_text[] = { + "Left", "Right" +}; + +static const struct soc_enum aifoutl_enum = + SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 7, 2, aif_text); + +static const struct snd_kcontrol_new aifoutl_mux = + SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum); + +static const struct soc_enum aifoutr_enum = + SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 6, 2, aif_text); + +static const struct snd_kcontrol_new aifoutr_mux = + SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum); + +static const struct soc_enum aifinl_enum = + SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 5, 2, aif_text); + +static const struct snd_kcontrol_new aifinl_mux = + SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum); + +static const struct soc_enum aifinr_enum = + SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 4, 2, aif_text); + +static const struct snd_kcontrol_new aifinr_mux = + SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum); + +static const struct snd_soc_dapm_widget wm8904_core_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", WM8904_CLOCK_RATES_2, 2, 0, sysclk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8904_CLOCK_RATES_2, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TOCLK", WM8904_CLOCK_RATES_2, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget wm8904_adc_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), + +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8904_MIC_BIAS_CONTROL_0, 0, 0), + +SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lin_mux), +SND_SOC_DAPM_MUX("Left Capture Inverting Mux", SND_SOC_NOPM, 0, 0, + &lin_inv_mux), +SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rin_mux), +SND_SOC_DAPM_MUX("Right Capture Inverting Mux", SND_SOC_NOPM, 0, 0, + &rin_inv_mux), + +SND_SOC_DAPM_PGA("Left Capture PGA", WM8904_POWER_MANAGEMENT_0, 1, 0, + NULL, 0), +SND_SOC_DAPM_PGA("Right Capture PGA", WM8904_POWER_MANAGEMENT_0, 0, 0, + NULL, 0), + +SND_SOC_DAPM_ADC("ADCL", NULL, WM8904_POWER_MANAGEMENT_6, 1, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, WM8904_POWER_MANAGEMENT_6, 0, 0), + +SND_SOC_DAPM_MUX("AIFOUTL Mux", SND_SOC_NOPM, 0, 0, &aifoutl_mux), +SND_SOC_DAPM_MUX("AIFOUTR Mux", SND_SOC_NOPM, 0, 0, &aifoutr_mux), + +SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_widget wm8904_dac_dapm_widgets[] = { +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &aifinl_mux), +SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &aifinr_mux), + +SND_SOC_DAPM_DAC("DACL", NULL, WM8904_POWER_MANAGEMENT_6, 3, 0), +SND_SOC_DAPM_DAC("DACR", NULL, WM8904_POWER_MANAGEMENT_6, 2, 0), + +SND_SOC_DAPM_SUPPLY("Charge pump", WM8904_CHARGE_PUMP_0, 0, 0, cp_event, + SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("HPL PGA", WM8904_POWER_MANAGEMENT_2, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("HPR PGA", WM8904_POWER_MANAGEMENT_2, 0, 0, NULL, 0), + +SND_SOC_DAPM_PGA("LINEL PGA", WM8904_POWER_MANAGEMENT_3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("LINER PGA", WM8904_POWER_MANAGEMENT_3, 0, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM, WM8904_ANALOGUE_HP_0, + 0, NULL, 0, out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_E("Line Output", SND_SOC_NOPM, WM8904_ANALOGUE_LINEOUT_0, + 0, NULL, 0, out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +SND_SOC_DAPM_OUTPUT("LINEOUTL"), +SND_SOC_DAPM_OUTPUT("LINEOUTR"), +}; + +static const char *out_mux_text[] = { + "DAC", "Bypass" +}; + +static const struct soc_enum hpl_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 3, 2, out_mux_text); + +static const struct snd_kcontrol_new hpl_mux = + SOC_DAPM_ENUM("HPL Mux", hpl_enum); + +static const struct soc_enum hpr_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 2, 2, out_mux_text); + +static const struct snd_kcontrol_new hpr_mux = + SOC_DAPM_ENUM("HPR Mux", hpr_enum); + +static const struct soc_enum linel_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 1, 2, out_mux_text); + +static const struct snd_kcontrol_new linel_mux = + SOC_DAPM_ENUM("LINEL Mux", linel_enum); + +static const struct soc_enum liner_enum = + SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); + +static const struct snd_kcontrol_new liner_mux = + SOC_DAPM_ENUM("LINEL Mux", liner_enum); + +static const char *sidetone_text[] = { + "None", "Left", "Right" +}; + +static const struct soc_enum dacl_sidetone_enum = + SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 2, 3, sidetone_text); + +static const struct snd_kcontrol_new dacl_sidetone_mux = + SOC_DAPM_ENUM("Left Sidetone Mux", dacl_sidetone_enum); + +static const struct soc_enum dacr_sidetone_enum = + SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 0, 3, sidetone_text); + +static const struct snd_kcontrol_new dacr_sidetone_mux = + SOC_DAPM_ENUM("Right Sidetone Mux", dacr_sidetone_enum); + +static const struct snd_soc_dapm_widget wm8904_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("Class G", WM8904_CLASS_W_0, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Bypass", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Bypass", SND_SOC_NOPM, 0, 0, NULL, 0), + +SND_SOC_DAPM_MUX("Left Sidetone", SND_SOC_NOPM, 0, 0, &dacl_sidetone_mux), +SND_SOC_DAPM_MUX("Right Sidetone", SND_SOC_NOPM, 0, 0, &dacr_sidetone_mux), + +SND_SOC_DAPM_MUX("HPL Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("HPR Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), +SND_SOC_DAPM_MUX("LINEL Mux", SND_SOC_NOPM, 0, 0, &linel_mux), +SND_SOC_DAPM_MUX("LINER Mux", SND_SOC_NOPM, 0, 0, &liner_mux), +}; + +static const struct snd_soc_dapm_route core_intercon[] = { + { "CLK_DSP", NULL, "SYSCLK" }, + { "TOCLK", NULL, "SYSCLK" }, +}; + +static const struct snd_soc_dapm_route adc_intercon[] = { + { "Left Capture Mux", "IN1L", "IN1L" }, + { "Left Capture Mux", "IN2L", "IN2L" }, + { "Left Capture Mux", "IN3L", "IN3L" }, + + { "Left Capture Inverting Mux", "IN1L", "IN1L" }, + { "Left Capture Inverting Mux", "IN2L", "IN2L" }, + { "Left Capture Inverting Mux", "IN3L", "IN3L" }, + + { "Right Capture Mux", "IN1R", "IN1R" }, + { "Right Capture Mux", "IN2R", "IN2R" }, + { "Right Capture Mux", "IN3R", "IN3R" }, + + { "Right Capture Inverting Mux", "IN1R", "IN1R" }, + { "Right Capture Inverting Mux", "IN2R", "IN2R" }, + { "Right Capture Inverting Mux", "IN3R", "IN3R" }, + + { "Left Capture PGA", NULL, "Left Capture Mux" }, + { "Left Capture PGA", NULL, "Left Capture Inverting Mux" }, + + { "Right Capture PGA", NULL, "Right Capture Mux" }, + { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, + + { "AIFOUTL", "Left", "ADCL" }, + { "AIFOUTL", "Right", "ADCR" }, + { "AIFOUTR", "Left", "ADCL" }, + { "AIFOUTR", "Right", "ADCR" }, + + { "ADCL", NULL, "CLK_DSP" }, + { "ADCL", NULL, "Left Capture PGA" }, + + { "ADCR", NULL, "CLK_DSP" }, + { "ADCR", NULL, "Right Capture PGA" }, +}; + +static const struct snd_soc_dapm_route dac_intercon[] = { + { "DACL", "Right", "AIFINR" }, + { "DACL", "Left", "AIFINL" }, + { "DACL", NULL, "CLK_DSP" }, + + { "DACR", "Right", "AIFINR" }, + { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "CLK_DSP" }, + + { "Charge pump", NULL, "SYSCLK" }, + + { "Headphone Output", NULL, "HPL PGA" }, + { "Headphone Output", NULL, "HPR PGA" }, + { "Headphone Output", NULL, "Charge pump" }, + { "Headphone Output", NULL, "TOCLK" }, + + { "Line Output", NULL, "LINEL PGA" }, + { "Line Output", NULL, "LINER PGA" }, + { "Line Output", NULL, "Charge pump" }, + { "Line Output", NULL, "TOCLK" }, + + { "HPOUTL", NULL, "Headphone Output" }, + { "HPOUTR", NULL, "Headphone Output" }, + + { "LINEOUTL", NULL, "Line Output" }, + { "LINEOUTR", NULL, "Line Output" }, +}; + +static const struct snd_soc_dapm_route wm8904_intercon[] = { + { "Left Sidetone", "Left", "ADCL" }, + { "Left Sidetone", "Right", "ADCR" }, + { "DACL", NULL, "Left Sidetone" }, + + { "Right Sidetone", "Left", "ADCL" }, + { "Right Sidetone", "Right", "ADCR" }, + { "DACR", NULL, "Right Sidetone" }, + + { "Left Bypass", NULL, "Class G" }, + { "Left Bypass", NULL, "Left Capture PGA" }, + + { "Right Bypass", NULL, "Class G" }, + { "Right Bypass", NULL, "Right Capture PGA" }, + + { "HPL Mux", "DAC", "DACL" }, + { "HPL Mux", "Bypass", "Left Bypass" }, + + { "HPR Mux", "DAC", "DACR" }, + { "HPR Mux", "Bypass", "Right Bypass" }, + + { "LINEL Mux", "DAC", "DACL" }, + { "LINEL Mux", "Bypass", "Left Bypass" }, + + { "LINER Mux", "DAC", "DACR" }, + { "LINER Mux", "Bypass", "Right Bypass" }, + + { "HPL PGA", NULL, "HPL Mux" }, + { "HPR PGA", NULL, "HPR Mux" }, + + { "LINEL PGA", NULL, "LINEL Mux" }, + { "LINER PGA", NULL, "LINER Mux" }, +}; + +static int wm8904_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_add_controls(codec, wm8904_adc_snd_controls, + ARRAY_SIZE(wm8904_adc_snd_controls)); + snd_soc_add_controls(codec, wm8904_dac_snd_controls, + ARRAY_SIZE(wm8904_dac_snd_controls)); + snd_soc_add_controls(codec, wm8904_snd_controls, + ARRAY_SIZE(wm8904_snd_controls)); + + snd_soc_dapm_new_controls(codec, wm8904_core_dapm_widgets, + ARRAY_SIZE(wm8904_core_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets, + ARRAY_SIZE(wm8904_adc_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, + ARRAY_SIZE(wm8904_dac_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets, + ARRAY_SIZE(wm8904_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, core_intercon, + ARRAY_SIZE(core_intercon)); + snd_soc_dapm_add_routes(codec, adc_intercon, ARRAY_SIZE(adc_intercon)); + snd_soc_dapm_add_routes(codec, dac_intercon, ARRAY_SIZE(dac_intercon)); + snd_soc_dapm_add_routes(codec, wm8904_intercon, + ARRAY_SIZE(wm8904_intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static struct { + int ratio; + unsigned int clk_sys_rate; +} clk_sys_rates[] = { + { 64, 0 }, + { 128, 1 }, + { 192, 2 }, + { 256, 3 }, + { 384, 4 }, + { 512, 5 }, + { 786, 6 }, + { 1024, 7 }, + { 1408, 8 }, + { 1536, 9 }, +}; + +static struct { + int rate; + int sample_rate; +} sample_rates[] = { + { 8000, 0 }, + { 11025, 1 }, + { 12000, 1 }, + { 16000, 2 }, + { 22050, 3 }, + { 24000, 3 }, + { 32000, 4 }, + { 44100, 5 }, + { 48000, 5 }, +}; + +static struct { + int div; /* *10 due to .5s */ + int bclk_div; +} bclk_divs[] = { + { 10, 0 }, + { 15, 1 }, + { 20, 2 }, + { 30, 3 }, + { 40, 4 }, + { 50, 5 }, + { 55, 6 }, + { 60, 7 }, + { 80, 8 }, + { 100, 9 }, + { 110, 10 }, + { 120, 11 }, + { 160, 12 }, + { 200, 13 }, + { 220, 14 }, + { 240, 16 }, + { 200, 17 }, + { 320, 18 }, + { 440, 19 }, + { 480, 20 }, +}; + + +static int wm8904_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8904_priv *wm8904 = codec->private_data; + int ret, i, best, best_val, cur_val; + unsigned int aif1 = 0; + unsigned int aif2 = 0; + unsigned int aif3 = 0; + unsigned int clock1 = 0; + unsigned int dac_digital1 = 0; + + /* What BCLK do we need? */ + wm8904->fs = params_rate(params); + if (wm8904->tdm_slots) { + dev_dbg(codec->dev, "Configuring for %d %d bit TDM slots\n", + wm8904->tdm_slots, wm8904->tdm_width); + wm8904->bclk = snd_soc_calc_bclk(wm8904->fs, + wm8904->tdm_width, 2, + wm8904->tdm_slots); + } else { + wm8904->bclk = snd_soc_params_to_bclk(params); + } + + dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm8904->bclk); + + ret = wm8904_configure_clocking(codec); + if (ret != 0) + return ret; + + /* Select nearest CLK_SYS_RATE */ + best = 0; + best_val = abs((wm8904->sysclk_rate / clk_sys_rates[0].ratio) + - wm8904->fs); + for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) { + cur_val = abs((wm8904->sysclk_rate / + clk_sys_rates[i].ratio) - wm8904->fs);; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n", + clk_sys_rates[best].ratio); + clock1 |= (clk_sys_rates[best].clk_sys_rate + << WM8904_CLK_SYS_RATE_SHIFT); + + /* SAMPLE_RATE */ + best = 0; + best_val = abs(wm8904->fs - sample_rates[0].rate); + for (i = 1; i < ARRAY_SIZE(sample_rates); i++) { + /* Closest match */ + cur_val = abs(wm8904->fs - sample_rates[i].rate); + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n", + sample_rates[best].rate); + clock1 |= (sample_rates[best].sample_rate + << WM8904_SAMPLE_RATE_SHIFT); + + /* Enable sloping stopband filter for low sample rates */ + if (wm8904->fs <= 24000) + dac_digital1 |= WM8904_DAC_SB_FILT; + + /* BCLK_DIV */ + best = 0; + best_val = INT_MAX; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = ((wm8904->sysclk_rate * 10) / bclk_divs[i].div) + - wm8904->bclk; + if (cur_val < 0) /* Table is sorted */ + break; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + wm8904->bclk = (wm8904->sysclk_rate * 10) / bclk_divs[best].div; + dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n", + bclk_divs[best].div, wm8904->bclk); + aif2 |= bclk_divs[best].bclk_div; + + /* LRCLK is a simple fraction of BCLK */ + dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm8904->bclk / wm8904->fs); + aif3 |= wm8904->bclk / wm8904->fs; + + /* Apply the settings */ + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1, + WM8904_DAC_SB_FILT, dac_digital1); + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1, + WM8904_AIF_WL_MASK, aif1); + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_2, + WM8904_BCLK_DIV_MASK, aif2); + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_3, + WM8904_LRCLK_RATE_MASK, aif3); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_1, + WM8904_SAMPLE_RATE_MASK | + WM8904_CLK_SYS_RATE_MASK, clock1); + + /* Update filters for the new settings */ + wm8904_set_retune_mobile(codec); + wm8904_set_deemph(codec); + + return 0; +} + + +static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8904_priv *priv = codec->private_data; + + switch (clk_id) { + case WM8904_CLK_MCLK: + priv->sysclk_src = clk_id; + priv->mclk_rate = freq; + break; + + case WM8904_CLK_FLL: + priv->sysclk_src = clk_id; + break; + + default: + return -EINVAL; + } + + dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq); + + wm8904_configure_clocking(codec); + + return 0; +} + +static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int aif1 = 0; + unsigned int aif3 = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif3 |= WM8904_LRCLK_DIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif1 |= WM8904_BCLK_DIR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif1 |= WM8904_BCLK_DIR; + aif3 |= WM8904_LRCLK_DIR; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif1 |= WM8904_AIF_LRCLK_INV; + case SND_SOC_DAIFMT_DSP_A: + aif1 |= 0x3; + break; + case SND_SOC_DAIFMT_I2S: + aif1 |= 0x2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif1 |= 0x1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8904_AIF_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif1 |= WM8904_AIF_BCLK_INV | WM8904_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8904_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif1 |= WM8904_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1, + WM8904_AIF_BCLK_INV | WM8904_AIF_LRCLK_INV | + WM8904_AIF_FMT_MASK | WM8904_BCLK_DIR, aif1); + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_3, + WM8904_LRCLK_DIR, aif3); + + return 0; +} + + +static int wm8904_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8904_priv *wm8904 = codec->private_data; + int aif1 = 0; + + /* Don't need to validate anything if we're turning off TDM */ + if (slots == 0) + goto out; + + /* Note that we allow configurations we can't handle ourselves - + * for example, we can generate clocks for slots 2 and up even if + * we can't use those slots ourselves. + */ + aif1 |= WM8904_AIFADC_TDM | WM8904_AIFDAC_TDM; + + switch (rx_mask) { + case 3: + break; + case 0xc: + aif1 |= WM8904_AIFADC_TDM_CHAN; + break; + default: + return -EINVAL; + } + + + switch (tx_mask) { + case 3: + break; + case 0xc: + aif1 |= WM8904_AIFDAC_TDM_CHAN; + break; + default: + return -EINVAL; + } + +out: + wm8904->tdm_width = slot_width; + wm8904->tdm_slots = slots / 2; + + snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1, + WM8904_AIFADC_TDM | WM8904_AIFADC_TDM_CHAN | + WM8904_AIFDAC_TDM | WM8904_AIFDAC_TDM_CHAN, aif1); + + return 0; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_clk_ref_div; + u16 n; + u16 k; +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + unsigned int div; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + fll_div->fll_clk_ref_div = 0; + while ((Fref / div) > 13500000) { + div *= 2; + fll_div->fll_clk_ref_div++; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + pr_debug("Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 4; + while (Fout * div < 90000000) { + div++; + if (div > 64) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + fll_div->fll_outdiv = div - 1; + + pr_debug("Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + target /= fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + /* Now, calculate N.K */ + Ndiv = target / Fref; + + fll_div->n = Ndiv; + Nmod = target % Fref; + pr_debug("Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, Fref); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + fll_div->k = K / 10; + + pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n", + fll_div->n, fll_div->k, + fll_div->fll_fratio, fll_div->fll_outdiv, + fll_div->fll_clk_ref_div); + + return 0; +} + +static int wm8904_set_fll(struct snd_soc_dai *dai, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8904_priv *wm8904 = codec->private_data; + struct _fll_div fll_div; + int ret, val; + int clock2, fll1; + + /* Any change? */ + if (source == wm8904->fll_src && Fref == wm8904->fll_fref && + Fout == wm8904->fll_fout) + return 0; + + if (Fout == 0) { + dev_dbg(codec->dev, "FLL disabled\n"); + + wm8904->fll_fref = 0; + wm8904->fll_fout = 0; + + /* Gate SYSCLK to avoid glitches */ + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_CLK_SYS_ENA, 0); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0); + + goto out; + } + + /* Validate the FLL ID */ + switch (source) { + case WM8904_FLL_MCLK: + case WM8904_FLL_LRCLK: + case WM8904_FLL_BCLK: + ret = fll_factors(&fll_div, Fref, Fout); + if (ret != 0) + return ret; + break; + + case WM8904_FLL_FREE_RUNNING: + dev_dbg(codec->dev, "Using free running FLL\n"); + /* Force 12MHz and output/4 for now */ + Fout = 12000000; + Fref = 12000000; + + memset(&fll_div, 0, sizeof(fll_div)); + fll_div.fll_outdiv = 3; + break; + + default: + dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id); + return -EINVAL; + } + + /* Save current state then disable the FLL and SYSCLK to avoid + * misclocking */ + clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2); + fll1 = snd_soc_read(codec, WM8904_FLL_CONTROL_1); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_CLK_SYS_ENA, 0); + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0); + + /* Unlock forced oscilator control to switch it on/off */ + snd_soc_update_bits(codec, WM8904_CONTROL_INTERFACE_TEST_1, + WM8904_USER_KEY, WM8904_USER_KEY); + + if (fll_id == WM8904_FLL_FREE_RUNNING) { + val = WM8904_FLL_FRC_NCO; + } else { + val = 0; + } + + snd_soc_update_bits(codec, WM8904_FLL_NCO_TEST_1, WM8904_FLL_FRC_NCO, + val); + snd_soc_update_bits(codec, WM8904_CONTROL_INTERFACE_TEST_1, + WM8904_USER_KEY, 0); + + switch (fll_id) { + case WM8904_FLL_MCLK: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5, + WM8904_FLL_CLK_REF_SRC_MASK, 0); + break; + + case WM8904_FLL_LRCLK: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5, + WM8904_FLL_CLK_REF_SRC_MASK, 1); + break; + + case WM8904_FLL_BCLK: + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5, + WM8904_FLL_CLK_REF_SRC_MASK, 2); + break; + } + + if (fll_div.k) + val = WM8904_FLL_FRACN_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_FRACN_ENA, val); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_2, + WM8904_FLL_OUTDIV_MASK | WM8904_FLL_FRATIO_MASK, + (fll_div.fll_outdiv << WM8904_FLL_OUTDIV_SHIFT) | + (fll_div.fll_fratio << WM8904_FLL_FRATIO_SHIFT)); + + snd_soc_write(codec, WM8904_FLL_CONTROL_3, fll_div.k); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_4, WM8904_FLL_N_MASK, + fll_div.n << WM8904_FLL_N_SHIFT); + + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5, + WM8904_FLL_CLK_REF_DIV_MASK, + fll_div.fll_clk_ref_div + << WM8904_FLL_CLK_REF_DIV_SHIFT); + + dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); + + wm8904->fll_fref = Fref; + wm8904->fll_fout = Fout; + wm8904->fll_src = source; + + /* Enable the FLL if it was previously active */ + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_OSC_ENA, fll1); + snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1, + WM8904_FLL_ENA, fll1); + +out: + /* Reenable SYSCLK if it was previously active */ + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, + WM8904_CLK_SYS_ENA, clock2); + + return 0; +} + +static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int val; + + if (mute) + val = WM8904_DAC_MUTE; + else + val = 0; + + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1, WM8904_DAC_MUTE, val); + + return 0; +} + +static int wm8904_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8904_priv *wm8904 = codec->private_data; + int ret, i; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID resistance 2*50k */ + snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0, + WM8904_VMID_RES_MASK, + 0x1 << WM8904_VMID_RES_SHIFT); + + /* Normal bias current */ + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_ISEL_MASK, 2 << WM8904_ISEL_SHIFT); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + /* Sync back cached values if they're + * different from the hardware default. + */ + for (i = 1; i < ARRAY_SIZE(wm8904->reg_cache); i++) { + if (!wm8904_access[i].writable) + continue; + + if (wm8904->reg_cache[i] == wm8904_reg[i]) + continue; + + snd_soc_write(codec, i, wm8904->reg_cache[i]); + } + + /* Enable bias */ + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_BIAS_ENA, WM8904_BIAS_ENA); + + /* Enable VMID, VMID buffering, 2*5k resistance */ + snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0, + WM8904_VMID_ENA | + WM8904_VMID_RES_MASK, + WM8904_VMID_ENA | + 0x3 << WM8904_VMID_RES_SHIFT); + + /* Let VMID ramp */ + msleep(1); + } + + /* Maintain VMID with 2*250k */ + snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0, + WM8904_VMID_RES_MASK, + 0x2 << WM8904_VMID_RES_SHIFT); + + /* Bias current *0.5 */ + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_ISEL_MASK, 0); + break; + + case SND_SOC_BIAS_OFF: + /* Turn off VMID */ + snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0, + WM8904_VMID_RES_MASK | WM8904_VMID_ENA, 0); + + /* Stop bias generation */ + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_BIAS_ENA, 0); + + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8904_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8904_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8904_dai_ops = { + .set_sysclk = wm8904_set_sysclk, + .set_fmt = wm8904_set_fmt, + .set_tdm_slot = wm8904_set_tdm_slot, + .set_pll = wm8904_set_fll, + .hw_params = wm8904_hw_params, + .digital_mute = wm8904_digital_mute, +}; + +struct snd_soc_dai wm8904_dai = { + .name = "WM8904", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8904_RATES, + .formats = WM8904_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8904_RATES, + .formats = WM8904_FORMATS, + }, + .ops = &wm8904_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8904_dai); + +#ifdef CONFIG_PM +static int wm8904_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8904_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8904_suspend NULL +#define wm8904_resume NULL +#endif + +static void wm8904_handle_retune_mobile_pdata(struct wm8904_priv *wm8904) +{ + struct snd_soc_codec *codec = &wm8904->codec; + struct wm8904_pdata *pdata = wm8904->pdata; + struct snd_kcontrol_new control = + SOC_ENUM_EXT("EQ Mode", + wm8904->retune_mobile_enum, + wm8904_get_retune_mobile_enum, + wm8904_put_retune_mobile_enum); + int ret, i, j; + const char **t; + + /* We need an array of texts for the enum API but the number + * of texts is likely to be less than the number of + * configurations due to the sample rate dependency of the + * configurations. */ + wm8904->num_retune_mobile_texts = 0; + wm8904->retune_mobile_texts = NULL; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + for (j = 0; j < wm8904->num_retune_mobile_texts; j++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8904->retune_mobile_texts[j]) == 0) + break; + } + + if (j != wm8904->num_retune_mobile_texts) + continue; + + /* Expand the array... */ + t = krealloc(wm8904->retune_mobile_texts, + sizeof(char *) * + (wm8904->num_retune_mobile_texts + 1), + GFP_KERNEL); + if (t == NULL) + continue; + + /* ...store the new entry... */ + t[wm8904->num_retune_mobile_texts] = + pdata->retune_mobile_cfgs[i].name; + + /* ...and remember the new version. */ + wm8904->num_retune_mobile_texts++; + wm8904->retune_mobile_texts = t; + } + + dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", + wm8904->num_retune_mobile_texts); + + wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts; + wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts; + + ret = snd_soc_add_controls(&wm8904->codec, &control, 1); + if (ret != 0) + dev_err(wm8904->codec.dev, + "Failed to add ReTune Mobile control: %d\n", ret); +} + +static void wm8904_handle_pdata(struct wm8904_priv *wm8904) +{ + struct snd_soc_codec *codec = &wm8904->codec; + struct wm8904_pdata *pdata = wm8904->pdata; + int ret, i; + + if (!pdata) { + snd_soc_add_controls(&wm8904->codec, wm8904_eq_controls, + ARRAY_SIZE(wm8904_eq_controls)); + return; + } + + dev_dbg(codec->dev, "%d DRC configurations\n", pdata->num_drc_cfgs); + + if (pdata->num_drc_cfgs) { + struct snd_kcontrol_new control = + SOC_ENUM_EXT("DRC Mode", wm8904->drc_enum, + wm8904_get_drc_enum, wm8904_put_drc_enum); + + /* We need an array of texts for the enum API */ + wm8904->drc_texts = kmalloc(sizeof(char *) + * pdata->num_drc_cfgs, GFP_KERNEL); + if (!wm8904->drc_texts) { + dev_err(wm8904->codec.dev, + "Failed to allocate %d DRC config texts\n", + pdata->num_drc_cfgs); + return; + } + + for (i = 0; i < pdata->num_drc_cfgs; i++) + wm8904->drc_texts[i] = pdata->drc_cfgs[i].name; + + wm8904->drc_enum.max = pdata->num_drc_cfgs; + wm8904->drc_enum.texts = wm8904->drc_texts; + + ret = snd_soc_add_controls(&wm8904->codec, &control, 1); + if (ret != 0) + dev_err(wm8904->codec.dev, + "Failed to add DRC mode control: %d\n", ret); + + wm8904_set_drc(codec); + } + + dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", + pdata->num_retune_mobile_cfgs); + + if (pdata->num_retune_mobile_cfgs) + wm8904_handle_retune_mobile_pdata(wm8904); + else + snd_soc_add_controls(&wm8904->codec, wm8904_eq_controls, + ARRAY_SIZE(wm8904_eq_controls)); +} + +static int wm8904_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8904_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8904_codec; + codec = wm8904_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + wm8904_handle_pdata(codec->private_data); + + wm8904_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +static int wm8904_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8904 = { + .probe = wm8904_probe, + .remove = wm8904_remove, + .suspend = wm8904_suspend, + .resume = wm8904_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8904); + +static int wm8904_register(struct wm8904_priv *wm8904, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8904->codec; + int i; + + if (wm8904_codec) { + dev_err(codec->dev, "Another WM8904 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8904; + codec->name = "WM8904"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8904_set_bias_level; + codec->dai = &wm8904_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8904_MAX_REGISTER; + codec->reg_cache = &wm8904->reg_cache; + codec->volatile_register = wm8904_volatile_register; + + memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++) + wm8904->supplies[i].supply = wm8904_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = snd_soc_read(codec, WM8904_SW_RESET_AND_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read ID register\n"); + goto err_enable; + } + if (ret != wm8904_reg[WM8904_SW_RESET_AND_ID]) { + dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret); + ret = -EINVAL; + goto err_enable; + } + + ret = snd_soc_read(codec, WM8904_REVISION); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_enable; + } + dev_info(codec->dev, "revision %c\n", ret + 'A'); + + ret = wm8904_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_enable; + } + + wm8904_dai.dev = codec->dev; + + /* Change some default settings - latch VU and enable ZC */ + wm8904->reg_cache[WM8904_ADC_DIGITAL_VOLUME_LEFT] |= WM8904_ADC_VU; + wm8904->reg_cache[WM8904_ADC_DIGITAL_VOLUME_RIGHT] |= WM8904_ADC_VU; + wm8904->reg_cache[WM8904_DAC_DIGITAL_VOLUME_LEFT] |= WM8904_DAC_VU; + wm8904->reg_cache[WM8904_DAC_DIGITAL_VOLUME_RIGHT] |= WM8904_DAC_VU; + wm8904->reg_cache[WM8904_ANALOGUE_OUT1_LEFT] |= WM8904_HPOUT_VU | + WM8904_HPOUTLZC; + wm8904->reg_cache[WM8904_ANALOGUE_OUT1_RIGHT] |= WM8904_HPOUT_VU | + WM8904_HPOUTRZC; + wm8904->reg_cache[WM8904_ANALOGUE_OUT2_LEFT] |= WM8904_LINEOUT_VU | + WM8904_LINEOUTLZC; + wm8904->reg_cache[WM8904_ANALOGUE_OUT2_RIGHT] |= WM8904_LINEOUT_VU | + WM8904_LINEOUTRZC; + wm8904->reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE; + + /* Set Class W by default - this will be managed by the Class + * G widget at runtime where bypass paths are available. + */ + wm8904->reg_cache[WM8904_CLASS_W_0] |= WM8904_CP_DYN_PWR; + + /* Use normal bias source */ + wm8904->reg_cache[WM8904_BIAS_CONTROL_0] &= ~WM8904_POBCTRL; + + wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + + wm8904_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8904_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); +err: + kfree(wm8904); + return ret; +} + +static void wm8904_unregister(struct wm8904_priv *wm8904) +{ + wm8904_set_bias_level(&wm8904->codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + snd_soc_unregister_dai(&wm8904_dai); + snd_soc_unregister_codec(&wm8904->codec); + kfree(wm8904); + wm8904_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8904_priv *wm8904; + struct snd_soc_codec *codec; + + wm8904 = kzalloc(sizeof(struct wm8904_priv), GFP_KERNEL); + if (wm8904 == NULL) + return -ENOMEM; + + codec = &wm8904->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8904); + codec->control_data = i2c; + wm8904->pdata = i2c->dev.platform_data; + + codec->dev = &i2c->dev; + + return wm8904_register(wm8904, SND_SOC_I2C); +} + +static __devexit int wm8904_i2c_remove(struct i2c_client *client) +{ + struct wm8904_priv *wm8904 = i2c_get_clientdata(client); + wm8904_unregister(wm8904); + return 0; +} + +static const struct i2c_device_id wm8904_i2c_id[] = { + { "wm8904", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); + +static struct i2c_driver wm8904_i2c_driver = { + .driver = { + .name = "WM8904", + .owner = THIS_MODULE, + }, + .probe = wm8904_i2c_probe, + .remove = __devexit_p(wm8904_i2c_remove), + .id_table = wm8904_i2c_id, +}; +#endif + +static int __init wm8904_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8904_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8904 I2C driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8904_modinit); + +static void __exit wm8904_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8904_i2c_driver); +#endif +} +module_exit(wm8904_exit); + +MODULE_DESCRIPTION("ASoC WM8904 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h new file mode 100644 index 0000000..b68886d --- /dev/null +++ b/sound/soc/codecs/wm8904.h @@ -0,0 +1,1681 @@ +/* + * wm8904.h -- WM8904 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics, plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8904_H +#define _WM8904_H + +#define WM8904_CLK_MCLK 1 +#define WM8904_CLK_FLL 2 + +#define WM8904_FLL_MCLK 1 +#define WM8904_FLL_BCLK 2 +#define WM8904_FLL_LRCLK 3 +#define WM8904_FLL_FREE_RUNNING 4 + +extern struct snd_soc_dai wm8904_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8904; + +/* + * Register values. + */ +#define WM8904_SW_RESET_AND_ID 0x00 +#define WM8904_REVISION 0x01 +#define WM8904_BIAS_CONTROL_0 0x04 +#define WM8904_VMID_CONTROL_0 0x05 +#define WM8904_MIC_BIAS_CONTROL_0 0x06 +#define WM8904_MIC_BIAS_CONTROL_1 0x07 +#define WM8904_ANALOGUE_DAC_0 0x08 +#define WM8904_MIC_FILTER_CONTROL 0x09 +#define WM8904_ANALOGUE_ADC_0 0x0A +#define WM8904_POWER_MANAGEMENT_0 0x0C +#define WM8904_POWER_MANAGEMENT_2 0x0E +#define WM8904_POWER_MANAGEMENT_3 0x0F +#define WM8904_POWER_MANAGEMENT_6 0x12 +#define WM8904_CLOCK_RATES_0 0x14 +#define WM8904_CLOCK_RATES_1 0x15 +#define WM8904_CLOCK_RATES_2 0x16 +#define WM8904_AUDIO_INTERFACE_0 0x18 +#define WM8904_AUDIO_INTERFACE_1 0x19 +#define WM8904_AUDIO_INTERFACE_2 0x1A +#define WM8904_AUDIO_INTERFACE_3 0x1B +#define WM8904_DAC_DIGITAL_VOLUME_LEFT 0x1E +#define WM8904_DAC_DIGITAL_VOLUME_RIGHT 0x1F +#define WM8904_DAC_DIGITAL_0 0x20 +#define WM8904_DAC_DIGITAL_1 0x21 +#define WM8904_ADC_DIGITAL_VOLUME_LEFT 0x24 +#define WM8904_ADC_DIGITAL_VOLUME_RIGHT 0x25 +#define WM8904_ADC_DIGITAL_0 0x26 +#define WM8904_DIGITAL_MICROPHONE_0 0x27 +#define WM8904_DRC_0 0x28 +#define WM8904_DRC_1 0x29 +#define WM8904_DRC_2 0x2A +#define WM8904_DRC_3 0x2B +#define WM8904_ANALOGUE_LEFT_INPUT_0 0x2C +#define WM8904_ANALOGUE_RIGHT_INPUT_0 0x2D +#define WM8904_ANALOGUE_LEFT_INPUT_1 0x2E +#define WM8904_ANALOGUE_RIGHT_INPUT_1 0x2F +#define WM8904_ANALOGUE_OUT1_LEFT 0x39 +#define WM8904_ANALOGUE_OUT1_RIGHT 0x3A +#define WM8904_ANALOGUE_OUT2_LEFT 0x3B +#define WM8904_ANALOGUE_OUT2_RIGHT 0x3C +#define WM8904_ANALOGUE_OUT12_ZC 0x3D +#define WM8904_DC_SERVO_0 0x43 +#define WM8904_DC_SERVO_1 0x44 +#define WM8904_DC_SERVO_2 0x45 +#define WM8904_DC_SERVO_4 0x47 +#define WM8904_DC_SERVO_5 0x48 +#define WM8904_DC_SERVO_6 0x49 +#define WM8904_DC_SERVO_7 0x4A +#define WM8904_DC_SERVO_8 0x4B +#define WM8904_DC_SERVO_9 0x4C +#define WM8904_DC_SERVO_READBACK_0 0x4D +#define WM8904_ANALOGUE_HP_0 0x5A +#define WM8904_ANALOGUE_LINEOUT_0 0x5E +#define WM8904_CHARGE_PUMP_0 0x62 +#define WM8904_CLASS_W_0 0x68 +#define WM8904_WRITE_SEQUENCER_0 0x6C +#define WM8904_WRITE_SEQUENCER_1 0x6D +#define WM8904_WRITE_SEQUENCER_2 0x6E +#define WM8904_WRITE_SEQUENCER_3 0x6F +#define WM8904_WRITE_SEQUENCER_4 0x70 +#define WM8904_FLL_CONTROL_1 0x74 +#define WM8904_FLL_CONTROL_2 0x75 +#define WM8904_FLL_CONTROL_3 0x76 +#define WM8904_FLL_CONTROL_4 0x77 +#define WM8904_FLL_CONTROL_5 0x78 +#define WM8904_GPIO_CONTROL_1 0x79 +#define WM8904_GPIO_CONTROL_2 0x7A +#define WM8904_GPIO_CONTROL_3 0x7B +#define WM8904_GPIO_CONTROL_4 0x7C +#define WM8904_DIGITAL_PULLS 0x7E +#define WM8904_INTERRUPT_STATUS 0x7F +#define WM8904_INTERRUPT_STATUS_MASK 0x80 +#define WM8904_INTERRUPT_POLARITY 0x81 +#define WM8904_INTERRUPT_DEBOUNCE 0x82 +#define WM8904_EQ1 0x86 +#define WM8904_EQ2 0x87 +#define WM8904_EQ3 0x88 +#define WM8904_EQ4 0x89 +#define WM8904_EQ5 0x8A +#define WM8904_EQ6 0x8B +#define WM8904_EQ7 0x8C +#define WM8904_EQ8 0x8D +#define WM8904_EQ9 0x8E +#define WM8904_EQ10 0x8F +#define WM8904_EQ11 0x90 +#define WM8904_EQ12 0x91 +#define WM8904_EQ13 0x92 +#define WM8904_EQ14 0x93 +#define WM8904_EQ15 0x94 +#define WM8904_EQ16 0x95 +#define WM8904_EQ17 0x96 +#define WM8904_EQ18 0x97 +#define WM8904_EQ19 0x98 +#define WM8904_EQ20 0x99 +#define WM8904_EQ21 0x9A +#define WM8904_EQ22 0x9B +#define WM8904_EQ23 0x9C +#define WM8904_EQ24 0x9D +#define WM8904_CONTROL_INTERFACE_TEST_1 0xA1 +#define WM8904_ANALOGUE_OUTPUT_BIAS_0 0xCC +#define WM8904_FLL_NCO_TEST_0 0xF7 +#define WM8904_FLL_NCO_TEST_1 0xF8 + +#define WM8904_REGISTER_COUNT 101 +#define WM8904_MAX_REGISTER 0xF8 + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - SW Reset and ID + */ +#define WM8904_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */ +#define WM8904_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */ +#define WM8904_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */ + +/* + * R1 (0x01) - Revision + */ +#define WM8904_REVISION_MASK 0x000F /* REVISION - [3:0] */ +#define WM8904_REVISION_SHIFT 0 /* REVISION - [3:0] */ +#define WM8904_REVISION_WIDTH 16 /* REVISION - [3:0] */ + +/* + * R4 (0x04) - Bias Control 0 + */ +#define WM8904_POBCTRL 0x0010 /* POBCTRL */ +#define WM8904_POBCTRL_MASK 0x0010 /* POBCTRL */ +#define WM8904_POBCTRL_SHIFT 4 /* POBCTRL */ +#define WM8904_POBCTRL_WIDTH 1 /* POBCTRL */ +#define WM8904_ISEL_MASK 0x000C /* ISEL - [3:2] */ +#define WM8904_ISEL_SHIFT 2 /* ISEL - [3:2] */ +#define WM8904_ISEL_WIDTH 2 /* ISEL - [3:2] */ +#define WM8904_STARTUP_BIAS_ENA 0x0002 /* STARTUP_BIAS_ENA */ +#define WM8904_STARTUP_BIAS_ENA_MASK 0x0002 /* STARTUP_BIAS_ENA */ +#define WM8904_STARTUP_BIAS_ENA_SHIFT 1 /* STARTUP_BIAS_ENA */ +#define WM8904_STARTUP_BIAS_ENA_WIDTH 1 /* STARTUP_BIAS_ENA */ +#define WM8904_BIAS_ENA 0x0001 /* BIAS_ENA */ +#define WM8904_BIAS_ENA_MASK 0x0001 /* BIAS_ENA */ +#define WM8904_BIAS_ENA_SHIFT 0 /* BIAS_ENA */ +#define WM8904_BIAS_ENA_WIDTH 1 /* BIAS_ENA */ + +/* + * R5 (0x05) - VMID Control 0 + */ +#define WM8904_VMID_BUF_ENA 0x0040 /* VMID_BUF_ENA */ +#define WM8904_VMID_BUF_ENA_MASK 0x0040 /* VMID_BUF_ENA */ +#define WM8904_VMID_BUF_ENA_SHIFT 6 /* VMID_BUF_ENA */ +#define WM8904_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */ +#define WM8904_VMID_RES_MASK 0x0006 /* VMID_RES - [2:1] */ +#define WM8904_VMID_RES_SHIFT 1 /* VMID_RES - [2:1] */ +#define WM8904_VMID_RES_WIDTH 2 /* VMID_RES - [2:1] */ +#define WM8904_VMID_ENA 0x0001 /* VMID_ENA */ +#define WM8904_VMID_ENA_MASK 0x0001 /* VMID_ENA */ +#define WM8904_VMID_ENA_SHIFT 0 /* VMID_ENA */ +#define WM8904_VMID_ENA_WIDTH 1 /* VMID_ENA */ + +/* + * R6 (0x06) - Mic Bias Control 0 + */ +#define WM8904_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */ +#define WM8904_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */ +#define WM8904_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */ +#define WM8904_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */ +#define WM8904_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */ +#define WM8904_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */ +#define WM8904_MICDET_ENA 0x0002 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_SHIFT 1 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_WIDTH 1 /* MICDET_ENA */ +#define WM8904_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */ + +/* + * R7 (0x07) - Mic Bias Control 1 + */ +#define WM8904_MIC_DET_FILTER_ENA 0x8000 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_MASK 0x8000 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_SHIFT 15 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_WIDTH 1 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA 0x4000 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_MASK 0x4000 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_SHIFT 14 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_WIDTH 1 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MICBIAS_SEL_MASK 0x0007 /* MICBIAS_SEL - [2:0] */ +#define WM8904_MICBIAS_SEL_SHIFT 0 /* MICBIAS_SEL - [2:0] */ +#define WM8904_MICBIAS_SEL_WIDTH 3 /* MICBIAS_SEL - [2:0] */ + +/* + * R8 (0x08) - Analogue DAC 0 + */ +#define WM8904_DAC_BIAS_SEL_MASK 0x0018 /* DAC_BIAS_SEL - [4:3] */ +#define WM8904_DAC_BIAS_SEL_SHIFT 3 /* DAC_BIAS_SEL - [4:3] */ +#define WM8904_DAC_BIAS_SEL_WIDTH 2 /* DAC_BIAS_SEL - [4:3] */ +#define WM8904_DAC_VMID_BIAS_SEL_MASK 0x0006 /* DAC_VMID_BIAS_SEL - [2:1] */ +#define WM8904_DAC_VMID_BIAS_SEL_SHIFT 1 /* DAC_VMID_BIAS_SEL - [2:1] */ +#define WM8904_DAC_VMID_BIAS_SEL_WIDTH 2 /* DAC_VMID_BIAS_SEL - [2:1] */ + +/* + * R9 (0x09) - mic Filter Control + */ +#define WM8904_MIC_DET_SET_THRESHOLD_MASK 0xF000 /* MIC_DET_SET_THRESHOLD - [15:12] */ +#define WM8904_MIC_DET_SET_THRESHOLD_SHIFT 12 /* MIC_DET_SET_THRESHOLD - [15:12] */ +#define WM8904_MIC_DET_SET_THRESHOLD_WIDTH 4 /* MIC_DET_SET_THRESHOLD - [15:12] */ +#define WM8904_MIC_DET_RESET_THRESHOLD_MASK 0x0F00 /* MIC_DET_RESET_THRESHOLD - [11:8] */ +#define WM8904_MIC_DET_RESET_THRESHOLD_SHIFT 8 /* MIC_DET_RESET_THRESHOLD - [11:8] */ +#define WM8904_MIC_DET_RESET_THRESHOLD_WIDTH 4 /* MIC_DET_RESET_THRESHOLD - [11:8] */ +#define WM8904_MIC_SHORT_SET_THRESHOLD_MASK 0x00F0 /* MIC_SHORT_SET_THRESHOLD - [7:4] */ +#define WM8904_MIC_SHORT_SET_THRESHOLD_SHIFT 4 /* MIC_SHORT_SET_THRESHOLD - [7:4] */ +#define WM8904_MIC_SHORT_SET_THRESHOLD_WIDTH 4 /* MIC_SHORT_SET_THRESHOLD - [7:4] */ +#define WM8904_MIC_SHORT_RESET_THRESHOLD_MASK 0x000F /* MIC_SHORT_RESET_THRESHOLD - [3:0] */ +#define WM8904_MIC_SHORT_RESET_THRESHOLD_SHIFT 0 /* MIC_SHORT_RESET_THRESHOLD - [3:0] */ +#define WM8904_MIC_SHORT_RESET_THRESHOLD_WIDTH 4 /* MIC_SHORT_RESET_THRESHOLD - [3:0] */ + +/* + * R10 (0x0A) - Analogue ADC 0 + */ +#define WM8904_ADC_OSR128 0x0001 /* ADC_OSR128 */ +#define WM8904_ADC_OSR128_MASK 0x0001 /* ADC_OSR128 */ +#define WM8904_ADC_OSR128_SHIFT 0 /* ADC_OSR128 */ +#define WM8904_ADC_OSR128_WIDTH 1 /* ADC_OSR128 */ + +/* + * R12 (0x0C) - Power Management 0 + */ +#define WM8904_INL_ENA 0x0002 /* INL_ENA */ +#define WM8904_INL_ENA_MASK 0x0002 /* INL_ENA */ +#define WM8904_INL_ENA_SHIFT 1 /* INL_ENA */ +#define WM8904_INL_ENA_WIDTH 1 /* INL_ENA */ +#define WM8904_INR_ENA 0x0001 /* INR_ENA */ +#define WM8904_INR_ENA_MASK 0x0001 /* INR_ENA */ +#define WM8904_INR_ENA_SHIFT 0 /* INR_ENA */ +#define WM8904_INR_ENA_WIDTH 1 /* INR_ENA */ + +/* + * R14 (0x0E) - Power Management 2 + */ +#define WM8904_HPL_PGA_ENA 0x0002 /* HPL_PGA_ENA */ +#define WM8904_HPL_PGA_ENA_MASK 0x0002 /* HPL_PGA_ENA */ +#define WM8904_HPL_PGA_ENA_SHIFT 1 /* HPL_PGA_ENA */ +#define WM8904_HPL_PGA_ENA_WIDTH 1 /* HPL_PGA_ENA */ +#define WM8904_HPR_PGA_ENA 0x0001 /* HPR_PGA_ENA */ +#define WM8904_HPR_PGA_ENA_MASK 0x0001 /* HPR_PGA_ENA */ +#define WM8904_HPR_PGA_ENA_SHIFT 0 /* HPR_PGA_ENA */ +#define WM8904_HPR_PGA_ENA_WIDTH 1 /* HPR_PGA_ENA */ + +/* + * R15 (0x0F) - Power Management 3 + */ +#define WM8904_LINEOUTL_PGA_ENA 0x0002 /* LINEOUTL_PGA_ENA */ +#define WM8904_LINEOUTL_PGA_ENA_MASK 0x0002 /* LINEOUTL_PGA_ENA */ +#define WM8904_LINEOUTL_PGA_ENA_SHIFT 1 /* LINEOUTL_PGA_ENA */ +#define WM8904_LINEOUTL_PGA_ENA_WIDTH 1 /* LINEOUTL_PGA_ENA */ +#define WM8904_LINEOUTR_PGA_ENA 0x0001 /* LINEOUTR_PGA_ENA */ +#define WM8904_LINEOUTR_PGA_ENA_MASK 0x0001 /* LINEOUTR_PGA_ENA */ +#define WM8904_LINEOUTR_PGA_ENA_SHIFT 0 /* LINEOUTR_PGA_ENA */ +#define WM8904_LINEOUTR_PGA_ENA_WIDTH 1 /* LINEOUTR_PGA_ENA */ + +/* + * R18 (0x12) - Power Management 6 + */ +#define WM8904_DACL_ENA 0x0008 /* DACL_ENA */ +#define WM8904_DACL_ENA_MASK 0x0008 /* DACL_ENA */ +#define WM8904_DACL_ENA_SHIFT 3 /* DACL_ENA */ +#define WM8904_DACL_ENA_WIDTH 1 /* DACL_ENA */ +#define WM8904_DACR_ENA 0x0004 /* DACR_ENA */ +#define WM8904_DACR_ENA_MASK 0x0004 /* DACR_ENA */ +#define WM8904_DACR_ENA_SHIFT 2 /* DACR_ENA */ +#define WM8904_DACR_ENA_WIDTH 1 /* DACR_ENA */ +#define WM8904_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8904_ADCL_ENA_MASK 0x0002 /* ADCL_ENA */ +#define WM8904_ADCL_ENA_SHIFT 1 /* ADCL_ENA */ +#define WM8904_ADCL_ENA_WIDTH 1 /* ADCL_ENA */ +#define WM8904_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8904_ADCR_ENA_MASK 0x0001 /* ADCR_ENA */ +#define WM8904_ADCR_ENA_SHIFT 0 /* ADCR_ENA */ +#define WM8904_ADCR_ENA_WIDTH 1 /* ADCR_ENA */ + +/* + * R20 (0x14) - Clock Rates 0 + */ +#define WM8904_TOCLK_RATE_DIV16 0x4000 /* TOCLK_RATE_DIV16 */ +#define WM8904_TOCLK_RATE_DIV16_MASK 0x4000 /* TOCLK_RATE_DIV16 */ +#define WM8904_TOCLK_RATE_DIV16_SHIFT 14 /* TOCLK_RATE_DIV16 */ +#define WM8904_TOCLK_RATE_DIV16_WIDTH 1 /* TOCLK_RATE_DIV16 */ +#define WM8904_TOCLK_RATE_X4 0x2000 /* TOCLK_RATE_X4 */ +#define WM8904_TOCLK_RATE_X4_MASK 0x2000 /* TOCLK_RATE_X4 */ +#define WM8904_TOCLK_RATE_X4_SHIFT 13 /* TOCLK_RATE_X4 */ +#define WM8904_TOCLK_RATE_X4_WIDTH 1 /* TOCLK_RATE_X4 */ +#define WM8904_SR_MODE 0x1000 /* SR_MODE */ +#define WM8904_SR_MODE_MASK 0x1000 /* SR_MODE */ +#define WM8904_SR_MODE_SHIFT 12 /* SR_MODE */ +#define WM8904_SR_MODE_WIDTH 1 /* SR_MODE */ +#define WM8904_MCLK_DIV 0x0001 /* MCLK_DIV */ +#define WM8904_MCLK_DIV_MASK 0x0001 /* MCLK_DIV */ +#define WM8904_MCLK_DIV_SHIFT 0 /* MCLK_DIV */ +#define WM8904_MCLK_DIV_WIDTH 1 /* MCLK_DIV */ + +/* + * R21 (0x15) - Clock Rates 1 + */ +#define WM8904_CLK_SYS_RATE_MASK 0x3C00 /* CLK_SYS_RATE - [13:10] */ +#define WM8904_CLK_SYS_RATE_SHIFT 10 /* CLK_SYS_RATE - [13:10] */ +#define WM8904_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [13:10] */ +#define WM8904_SAMPLE_RATE_MASK 0x0007 /* SAMPLE_RATE - [2:0] */ +#define WM8904_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [2:0] */ +#define WM8904_SAMPLE_RATE_WIDTH 3 /* SAMPLE_RATE - [2:0] */ + +/* + * R22 (0x16) - Clock Rates 2 + */ +#define WM8904_MCLK_INV 0x8000 /* MCLK_INV */ +#define WM8904_MCLK_INV_MASK 0x8000 /* MCLK_INV */ +#define WM8904_MCLK_INV_SHIFT 15 /* MCLK_INV */ +#define WM8904_MCLK_INV_WIDTH 1 /* MCLK_INV */ +#define WM8904_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8904_SYSCLK_SRC_MASK 0x4000 /* SYSCLK_SRC */ +#define WM8904_SYSCLK_SRC_SHIFT 14 /* SYSCLK_SRC */ +#define WM8904_SYSCLK_SRC_WIDTH 1 /* SYSCLK_SRC */ +#define WM8904_TOCLK_RATE 0x1000 /* TOCLK_RATE */ +#define WM8904_TOCLK_RATE_MASK 0x1000 /* TOCLK_RATE */ +#define WM8904_TOCLK_RATE_SHIFT 12 /* TOCLK_RATE */ +#define WM8904_TOCLK_RATE_WIDTH 1 /* TOCLK_RATE */ +#define WM8904_OPCLK_ENA 0x0008 /* OPCLK_ENA */ +#define WM8904_OPCLK_ENA_MASK 0x0008 /* OPCLK_ENA */ +#define WM8904_OPCLK_ENA_SHIFT 3 /* OPCLK_ENA */ +#define WM8904_OPCLK_ENA_WIDTH 1 /* OPCLK_ENA */ +#define WM8904_CLK_SYS_ENA 0x0004 /* CLK_SYS_ENA */ +#define WM8904_CLK_SYS_ENA_MASK 0x0004 /* CLK_SYS_ENA */ +#define WM8904_CLK_SYS_ENA_SHIFT 2 /* CLK_SYS_ENA */ +#define WM8904_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */ +#define WM8904_CLK_DSP_ENA 0x0002 /* CLK_DSP_ENA */ +#define WM8904_CLK_DSP_ENA_MASK 0x0002 /* CLK_DSP_ENA */ +#define WM8904_CLK_DSP_ENA_SHIFT 1 /* CLK_DSP_ENA */ +#define WM8904_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */ +#define WM8904_TOCLK_ENA 0x0001 /* TOCLK_ENA */ +#define WM8904_TOCLK_ENA_MASK 0x0001 /* TOCLK_ENA */ +#define WM8904_TOCLK_ENA_SHIFT 0 /* TOCLK_ENA */ +#define WM8904_TOCLK_ENA_WIDTH 1 /* TOCLK_ENA */ + +/* + * R24 (0x18) - Audio Interface 0 + */ +#define WM8904_DACL_DATINV 0x1000 /* DACL_DATINV */ +#define WM8904_DACL_DATINV_MASK 0x1000 /* DACL_DATINV */ +#define WM8904_DACL_DATINV_SHIFT 12 /* DACL_DATINV */ +#define WM8904_DACL_DATINV_WIDTH 1 /* DACL_DATINV */ +#define WM8904_DACR_DATINV 0x0800 /* DACR_DATINV */ +#define WM8904_DACR_DATINV_MASK 0x0800 /* DACR_DATINV */ +#define WM8904_DACR_DATINV_SHIFT 11 /* DACR_DATINV */ +#define WM8904_DACR_DATINV_WIDTH 1 /* DACR_DATINV */ +#define WM8904_DAC_BOOST_MASK 0x0600 /* DAC_BOOST - [10:9] */ +#define WM8904_DAC_BOOST_SHIFT 9 /* DAC_BOOST - [10:9] */ +#define WM8904_DAC_BOOST_WIDTH 2 /* DAC_BOOST - [10:9] */ +#define WM8904_LOOPBACK 0x0100 /* LOOPBACK */ +#define WM8904_LOOPBACK_MASK 0x0100 /* LOOPBACK */ +#define WM8904_LOOPBACK_SHIFT 8 /* LOOPBACK */ +#define WM8904_LOOPBACK_WIDTH 1 /* LOOPBACK */ +#define WM8904_AIFADCL_SRC 0x0080 /* AIFADCL_SRC */ +#define WM8904_AIFADCL_SRC_MASK 0x0080 /* AIFADCL_SRC */ +#define WM8904_AIFADCL_SRC_SHIFT 7 /* AIFADCL_SRC */ +#define WM8904_AIFADCL_SRC_WIDTH 1 /* AIFADCL_SRC */ +#define WM8904_AIFADCR_SRC 0x0040 /* AIFADCR_SRC */ +#define WM8904_AIFADCR_SRC_MASK 0x0040 /* AIFADCR_SRC */ +#define WM8904_AIFADCR_SRC_SHIFT 6 /* AIFADCR_SRC */ +#define WM8904_AIFADCR_SRC_WIDTH 1 /* AIFADCR_SRC */ +#define WM8904_AIFDACL_SRC 0x0020 /* AIFDACL_SRC */ +#define WM8904_AIFDACL_SRC_MASK 0x0020 /* AIFDACL_SRC */ +#define WM8904_AIFDACL_SRC_SHIFT 5 /* AIFDACL_SRC */ +#define WM8904_AIFDACL_SRC_WIDTH 1 /* AIFDACL_SRC */ +#define WM8904_AIFDACR_SRC 0x0010 /* AIFDACR_SRC */ +#define WM8904_AIFDACR_SRC_MASK 0x0010 /* AIFDACR_SRC */ +#define WM8904_AIFDACR_SRC_SHIFT 4 /* AIFDACR_SRC */ +#define WM8904_AIFDACR_SRC_WIDTH 1 /* AIFDACR_SRC */ +#define WM8904_ADC_COMP 0x0008 /* ADC_COMP */ +#define WM8904_ADC_COMP_MASK 0x0008 /* ADC_COMP */ +#define WM8904_ADC_COMP_SHIFT 3 /* ADC_COMP */ +#define WM8904_ADC_COMP_WIDTH 1 /* ADC_COMP */ +#define WM8904_ADC_COMPMODE 0x0004 /* ADC_COMPMODE */ +#define WM8904_ADC_COMPMODE_MASK 0x0004 /* ADC_COMPMODE */ +#define WM8904_ADC_COMPMODE_SHIFT 2 /* ADC_COMPMODE */ +#define WM8904_ADC_COMPMODE_WIDTH 1 /* ADC_COMPMODE */ +#define WM8904_DAC_COMP 0x0002 /* DAC_COMP */ +#define WM8904_DAC_COMP_MASK 0x0002 /* DAC_COMP */ +#define WM8904_DAC_COMP_SHIFT 1 /* DAC_COMP */ +#define WM8904_DAC_COMP_WIDTH 1 /* DAC_COMP */ +#define WM8904_DAC_COMPMODE 0x0001 /* DAC_COMPMODE */ +#define WM8904_DAC_COMPMODE_MASK 0x0001 /* DAC_COMPMODE */ +#define WM8904_DAC_COMPMODE_SHIFT 0 /* DAC_COMPMODE */ +#define WM8904_DAC_COMPMODE_WIDTH 1 /* DAC_COMPMODE */ + +/* + * R25 (0x19) - Audio Interface 1 + */ +#define WM8904_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8904_AIFDAC_TDM_MASK 0x2000 /* AIFDAC_TDM */ +#define WM8904_AIFDAC_TDM_SHIFT 13 /* AIFDAC_TDM */ +#define WM8904_AIFDAC_TDM_WIDTH 1 /* AIFDAC_TDM */ +#define WM8904_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8904_AIFDAC_TDM_CHAN_MASK 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8904_AIFDAC_TDM_CHAN_SHIFT 12 /* AIFDAC_TDM_CHAN */ +#define WM8904_AIFDAC_TDM_CHAN_WIDTH 1 /* AIFDAC_TDM_CHAN */ +#define WM8904_AIFADC_TDM 0x0800 /* AIFADC_TDM */ +#define WM8904_AIFADC_TDM_MASK 0x0800 /* AIFADC_TDM */ +#define WM8904_AIFADC_TDM_SHIFT 11 /* AIFADC_TDM */ +#define WM8904_AIFADC_TDM_WIDTH 1 /* AIFADC_TDM */ +#define WM8904_AIFADC_TDM_CHAN 0x0400 /* AIFADC_TDM_CHAN */ +#define WM8904_AIFADC_TDM_CHAN_MASK 0x0400 /* AIFADC_TDM_CHAN */ +#define WM8904_AIFADC_TDM_CHAN_SHIFT 10 /* AIFADC_TDM_CHAN */ +#define WM8904_AIFADC_TDM_CHAN_WIDTH 1 /* AIFADC_TDM_CHAN */ +#define WM8904_AIF_TRIS 0x0100 /* AIF_TRIS */ +#define WM8904_AIF_TRIS_MASK 0x0100 /* AIF_TRIS */ +#define WM8904_AIF_TRIS_SHIFT 8 /* AIF_TRIS */ +#define WM8904_AIF_TRIS_WIDTH 1 /* AIF_TRIS */ +#define WM8904_AIF_BCLK_INV 0x0080 /* AIF_BCLK_INV */ +#define WM8904_AIF_BCLK_INV_MASK 0x0080 /* AIF_BCLK_INV */ +#define WM8904_AIF_BCLK_INV_SHIFT 7 /* AIF_BCLK_INV */ +#define WM8904_AIF_BCLK_INV_WIDTH 1 /* AIF_BCLK_INV */ +#define WM8904_BCLK_DIR 0x0040 /* BCLK_DIR */ +#define WM8904_BCLK_DIR_MASK 0x0040 /* BCLK_DIR */ +#define WM8904_BCLK_DIR_SHIFT 6 /* BCLK_DIR */ +#define WM8904_BCLK_DIR_WIDTH 1 /* BCLK_DIR */ +#define WM8904_AIF_LRCLK_INV 0x0010 /* AIF_LRCLK_INV */ +#define WM8904_AIF_LRCLK_INV_MASK 0x0010 /* AIF_LRCLK_INV */ +#define WM8904_AIF_LRCLK_INV_SHIFT 4 /* AIF_LRCLK_INV */ +#define WM8904_AIF_LRCLK_INV_WIDTH 1 /* AIF_LRCLK_INV */ +#define WM8904_AIF_WL_MASK 0x000C /* AIF_WL - [3:2] */ +#define WM8904_AIF_WL_SHIFT 2 /* AIF_WL - [3:2] */ +#define WM8904_AIF_WL_WIDTH 2 /* AIF_WL - [3:2] */ +#define WM8904_AIF_FMT_MASK 0x0003 /* AIF_FMT - [1:0] */ +#define WM8904_AIF_FMT_SHIFT 0 /* AIF_FMT - [1:0] */ +#define WM8904_AIF_FMT_WIDTH 2 /* AIF_FMT - [1:0] */ + +/* + * R26 (0x1A) - Audio Interface 2 + */ +#define WM8904_OPCLK_DIV_MASK 0x0F00 /* OPCLK_DIV - [11:8] */ +#define WM8904_OPCLK_DIV_SHIFT 8 /* OPCLK_DIV - [11:8] */ +#define WM8904_OPCLK_DIV_WIDTH 4 /* OPCLK_DIV - [11:8] */ +#define WM8904_BCLK_DIV_MASK 0x001F /* BCLK_DIV - [4:0] */ +#define WM8904_BCLK_DIV_SHIFT 0 /* BCLK_DIV - [4:0] */ +#define WM8904_BCLK_DIV_WIDTH 5 /* BCLK_DIV - [4:0] */ + +/* + * R27 (0x1B) - Audio Interface 3 + */ +#define WM8904_LRCLK_DIR 0x0800 /* LRCLK_DIR */ +#define WM8904_LRCLK_DIR_MASK 0x0800 /* LRCLK_DIR */ +#define WM8904_LRCLK_DIR_SHIFT 11 /* LRCLK_DIR */ +#define WM8904_LRCLK_DIR_WIDTH 1 /* LRCLK_DIR */ +#define WM8904_LRCLK_RATE_MASK 0x07FF /* LRCLK_RATE - [10:0] */ +#define WM8904_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [10:0] */ +#define WM8904_LRCLK_RATE_WIDTH 11 /* LRCLK_RATE - [10:0] */ + +/* + * R30 (0x1E) - DAC Digital Volume Left + */ +#define WM8904_DAC_VU 0x0100 /* DAC_VU */ +#define WM8904_DAC_VU_MASK 0x0100 /* DAC_VU */ +#define WM8904_DAC_VU_SHIFT 8 /* DAC_VU */ +#define WM8904_DAC_VU_WIDTH 1 /* DAC_VU */ +#define WM8904_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8904_DACL_VOL_SHIFT 0 /* DACL_VOL - [7:0] */ +#define WM8904_DACL_VOL_WIDTH 8 /* DACL_VOL - [7:0] */ + +/* + * R31 (0x1F) - DAC Digital Volume Right + */ +#define WM8904_DAC_VU 0x0100 /* DAC_VU */ +#define WM8904_DAC_VU_MASK 0x0100 /* DAC_VU */ +#define WM8904_DAC_VU_SHIFT 8 /* DAC_VU */ +#define WM8904_DAC_VU_WIDTH 1 /* DAC_VU */ +#define WM8904_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8904_DACR_VOL_SHIFT 0 /* DACR_VOL - [7:0] */ +#define WM8904_DACR_VOL_WIDTH 8 /* DACR_VOL - [7:0] */ + +/* + * R32 (0x20) - DAC Digital 0 + */ +#define WM8904_ADCL_DAC_SVOL_MASK 0x0F00 /* ADCL_DAC_SVOL - [11:8] */ +#define WM8904_ADCL_DAC_SVOL_SHIFT 8 /* ADCL_DAC_SVOL - [11:8] */ +#define WM8904_ADCL_DAC_SVOL_WIDTH 4 /* ADCL_DAC_SVOL - [11:8] */ +#define WM8904_ADCR_DAC_SVOL_MASK 0x00F0 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8904_ADCR_DAC_SVOL_SHIFT 4 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8904_ADCR_DAC_SVOL_WIDTH 4 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8904_ADC_TO_DACL_MASK 0x000C /* ADC_TO_DACL - [3:2] */ +#define WM8904_ADC_TO_DACL_SHIFT 2 /* ADC_TO_DACL - [3:2] */ +#define WM8904_ADC_TO_DACL_WIDTH 2 /* ADC_TO_DACL - [3:2] */ +#define WM8904_ADC_TO_DACR_MASK 0x0003 /* ADC_TO_DACR - [1:0] */ +#define WM8904_ADC_TO_DACR_SHIFT 0 /* ADC_TO_DACR - [1:0] */ +#define WM8904_ADC_TO_DACR_WIDTH 2 /* ADC_TO_DACR - [1:0] */ + +/* + * R33 (0x21) - DAC Digital 1 + */ +#define WM8904_DAC_MONO 0x1000 /* DAC_MONO */ +#define WM8904_DAC_MONO_MASK 0x1000 /* DAC_MONO */ +#define WM8904_DAC_MONO_SHIFT 12 /* DAC_MONO */ +#define WM8904_DAC_MONO_WIDTH 1 /* DAC_MONO */ +#define WM8904_DAC_SB_FILT 0x0800 /* DAC_SB_FILT */ +#define WM8904_DAC_SB_FILT_MASK 0x0800 /* DAC_SB_FILT */ +#define WM8904_DAC_SB_FILT_SHIFT 11 /* DAC_SB_FILT */ +#define WM8904_DAC_SB_FILT_WIDTH 1 /* DAC_SB_FILT */ +#define WM8904_DAC_MUTERATE 0x0400 /* DAC_MUTERATE */ +#define WM8904_DAC_MUTERATE_MASK 0x0400 /* DAC_MUTERATE */ +#define WM8904_DAC_MUTERATE_SHIFT 10 /* DAC_MUTERATE */ +#define WM8904_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */ +#define WM8904_DAC_UNMUTE_RAMP 0x0200 /* DAC_UNMUTE_RAMP */ +#define WM8904_DAC_UNMUTE_RAMP_MASK 0x0200 /* DAC_UNMUTE_RAMP */ +#define WM8904_DAC_UNMUTE_RAMP_SHIFT 9 /* DAC_UNMUTE_RAMP */ +#define WM8904_DAC_UNMUTE_RAMP_WIDTH 1 /* DAC_UNMUTE_RAMP */ +#define WM8904_DAC_OSR128 0x0040 /* DAC_OSR128 */ +#define WM8904_DAC_OSR128_MASK 0x0040 /* DAC_OSR128 */ +#define WM8904_DAC_OSR128_SHIFT 6 /* DAC_OSR128 */ +#define WM8904_DAC_OSR128_WIDTH 1 /* DAC_OSR128 */ +#define WM8904_DAC_MUTE 0x0008 /* DAC_MUTE */ +#define WM8904_DAC_MUTE_MASK 0x0008 /* DAC_MUTE */ +#define WM8904_DAC_MUTE_SHIFT 3 /* DAC_MUTE */ +#define WM8904_DAC_MUTE_WIDTH 1 /* DAC_MUTE */ +#define WM8904_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */ +#define WM8904_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */ +#define WM8904_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */ + +/* + * R36 (0x24) - ADC Digital Volume Left + */ +#define WM8904_ADC_VU 0x0100 /* ADC_VU */ +#define WM8904_ADC_VU_MASK 0x0100 /* ADC_VU */ +#define WM8904_ADC_VU_SHIFT 8 /* ADC_VU */ +#define WM8904_ADC_VU_WIDTH 1 /* ADC_VU */ +#define WM8904_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8904_ADCL_VOL_SHIFT 0 /* ADCL_VOL - [7:0] */ +#define WM8904_ADCL_VOL_WIDTH 8 /* ADCL_VOL - [7:0] */ + +/* + * R37 (0x25) - ADC Digital Volume Right + */ +#define WM8904_ADC_VU 0x0100 /* ADC_VU */ +#define WM8904_ADC_VU_MASK 0x0100 /* ADC_VU */ +#define WM8904_ADC_VU_SHIFT 8 /* ADC_VU */ +#define WM8904_ADC_VU_WIDTH 1 /* ADC_VU */ +#define WM8904_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8904_ADCR_VOL_SHIFT 0 /* ADCR_VOL - [7:0] */ +#define WM8904_ADCR_VOL_WIDTH 8 /* ADCR_VOL - [7:0] */ + +/* + * R38 (0x26) - ADC Digital 0 + */ +#define WM8904_ADC_HPF_CUT_MASK 0x0060 /* ADC_HPF_CUT - [6:5] */ +#define WM8904_ADC_HPF_CUT_SHIFT 5 /* ADC_HPF_CUT - [6:5] */ +#define WM8904_ADC_HPF_CUT_WIDTH 2 /* ADC_HPF_CUT - [6:5] */ +#define WM8904_ADC_HPF 0x0010 /* ADC_HPF */ +#define WM8904_ADC_HPF_MASK 0x0010 /* ADC_HPF */ +#define WM8904_ADC_HPF_SHIFT 4 /* ADC_HPF */ +#define WM8904_ADC_HPF_WIDTH 1 /* ADC_HPF */ +#define WM8904_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8904_ADCL_DATINV_MASK 0x0002 /* ADCL_DATINV */ +#define WM8904_ADCL_DATINV_SHIFT 1 /* ADCL_DATINV */ +#define WM8904_ADCL_DATINV_WIDTH 1 /* ADCL_DATINV */ +#define WM8904_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8904_ADCR_DATINV_MASK 0x0001 /* ADCR_DATINV */ +#define WM8904_ADCR_DATINV_SHIFT 0 /* ADCR_DATINV */ +#define WM8904_ADCR_DATINV_WIDTH 1 /* ADCR_DATINV */ + +/* + * R39 (0x27) - Digital Microphone 0 + */ +#define WM8904_DMIC_ENA 0x1000 /* DMIC_ENA */ +#define WM8904_DMIC_ENA_MASK 0x1000 /* DMIC_ENA */ +#define WM8904_DMIC_ENA_SHIFT 12 /* DMIC_ENA */ +#define WM8904_DMIC_ENA_WIDTH 1 /* DMIC_ENA */ +#define WM8904_DMIC_SRC 0x0800 /* DMIC_SRC */ +#define WM8904_DMIC_SRC_MASK 0x0800 /* DMIC_SRC */ +#define WM8904_DMIC_SRC_SHIFT 11 /* DMIC_SRC */ +#define WM8904_DMIC_SRC_WIDTH 1 /* DMIC_SRC */ + +/* + * R40 (0x28) - DRC 0 + */ +#define WM8904_DRC_ENA 0x8000 /* DRC_ENA */ +#define WM8904_DRC_ENA_MASK 0x8000 /* DRC_ENA */ +#define WM8904_DRC_ENA_SHIFT 15 /* DRC_ENA */ +#define WM8904_DRC_ENA_WIDTH 1 /* DRC_ENA */ +#define WM8904_DRC_DAC_PATH 0x4000 /* DRC_DAC_PATH */ +#define WM8904_DRC_DAC_PATH_MASK 0x4000 /* DRC_DAC_PATH */ +#define WM8904_DRC_DAC_PATH_SHIFT 14 /* DRC_DAC_PATH */ +#define WM8904_DRC_DAC_PATH_WIDTH 1 /* DRC_DAC_PATH */ +#define WM8904_DRC_GS_HYST_LVL_MASK 0x1800 /* DRC_GS_HYST_LVL - [12:11] */ +#define WM8904_DRC_GS_HYST_LVL_SHIFT 11 /* DRC_GS_HYST_LVL - [12:11] */ +#define WM8904_DRC_GS_HYST_LVL_WIDTH 2 /* DRC_GS_HYST_LVL - [12:11] */ +#define WM8904_DRC_STARTUP_GAIN_MASK 0x07C0 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM8904_DRC_STARTUP_GAIN_SHIFT 6 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM8904_DRC_STARTUP_GAIN_WIDTH 5 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM8904_DRC_FF_DELAY 0x0020 /* DRC_FF_DELAY */ +#define WM8904_DRC_FF_DELAY_MASK 0x0020 /* DRC_FF_DELAY */ +#define WM8904_DRC_FF_DELAY_SHIFT 5 /* DRC_FF_DELAY */ +#define WM8904_DRC_FF_DELAY_WIDTH 1 /* DRC_FF_DELAY */ +#define WM8904_DRC_GS_ENA 0x0008 /* DRC_GS_ENA */ +#define WM8904_DRC_GS_ENA_MASK 0x0008 /* DRC_GS_ENA */ +#define WM8904_DRC_GS_ENA_SHIFT 3 /* DRC_GS_ENA */ +#define WM8904_DRC_GS_ENA_WIDTH 1 /* DRC_GS_ENA */ +#define WM8904_DRC_QR 0x0004 /* DRC_QR */ +#define WM8904_DRC_QR_MASK 0x0004 /* DRC_QR */ +#define WM8904_DRC_QR_SHIFT 2 /* DRC_QR */ +#define WM8904_DRC_QR_WIDTH 1 /* DRC_QR */ +#define WM8904_DRC_ANTICLIP 0x0002 /* DRC_ANTICLIP */ +#define WM8904_DRC_ANTICLIP_MASK 0x0002 /* DRC_ANTICLIP */ +#define WM8904_DRC_ANTICLIP_SHIFT 1 /* DRC_ANTICLIP */ +#define WM8904_DRC_ANTICLIP_WIDTH 1 /* DRC_ANTICLIP */ +#define WM8904_DRC_GS_HYST 0x0001 /* DRC_GS_HYST */ +#define WM8904_DRC_GS_HYST_MASK 0x0001 /* DRC_GS_HYST */ +#define WM8904_DRC_GS_HYST_SHIFT 0 /* DRC_GS_HYST */ +#define WM8904_DRC_GS_HYST_WIDTH 1 /* DRC_GS_HYST */ + +/* + * R41 (0x29) - DRC 1 + */ +#define WM8904_DRC_ATK_MASK 0xF000 /* DRC_ATK - [15:12] */ +#define WM8904_DRC_ATK_SHIFT 12 /* DRC_ATK - [15:12] */ +#define WM8904_DRC_ATK_WIDTH 4 /* DRC_ATK - [15:12] */ +#define WM8904_DRC_DCY_MASK 0x0F00 /* DRC_DCY - [11:8] */ +#define WM8904_DRC_DCY_SHIFT 8 /* DRC_DCY - [11:8] */ +#define WM8904_DRC_DCY_WIDTH 4 /* DRC_DCY - [11:8] */ +#define WM8904_DRC_QR_THR_MASK 0x00C0 /* DRC_QR_THR - [7:6] */ +#define WM8904_DRC_QR_THR_SHIFT 6 /* DRC_QR_THR - [7:6] */ +#define WM8904_DRC_QR_THR_WIDTH 2 /* DRC_QR_THR - [7:6] */ +#define WM8904_DRC_QR_DCY_MASK 0x0030 /* DRC_QR_DCY - [5:4] */ +#define WM8904_DRC_QR_DCY_SHIFT 4 /* DRC_QR_DCY - [5:4] */ +#define WM8904_DRC_QR_DCY_WIDTH 2 /* DRC_QR_DCY - [5:4] */ +#define WM8904_DRC_MINGAIN_MASK 0x000C /* DRC_MINGAIN - [3:2] */ +#define WM8904_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [3:2] */ +#define WM8904_DRC_MINGAIN_WIDTH 2 /* DRC_MINGAIN - [3:2] */ +#define WM8904_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */ +#define WM8904_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */ +#define WM8904_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */ + +/* + * R42 (0x2A) - DRC 2 + */ +#define WM8904_DRC_HI_COMP_MASK 0x0038 /* DRC_HI_COMP - [5:3] */ +#define WM8904_DRC_HI_COMP_SHIFT 3 /* DRC_HI_COMP - [5:3] */ +#define WM8904_DRC_HI_COMP_WIDTH 3 /* DRC_HI_COMP - [5:3] */ +#define WM8904_DRC_LO_COMP_MASK 0x0007 /* DRC_LO_COMP - [2:0] */ +#define WM8904_DRC_LO_COMP_SHIFT 0 /* DRC_LO_COMP - [2:0] */ +#define WM8904_DRC_LO_COMP_WIDTH 3 /* DRC_LO_COMP - [2:0] */ + +/* + * R43 (0x2B) - DRC 3 + */ +#define WM8904_DRC_KNEE_IP_MASK 0x07E0 /* DRC_KNEE_IP - [10:5] */ +#define WM8904_DRC_KNEE_IP_SHIFT 5 /* DRC_KNEE_IP - [10:5] */ +#define WM8904_DRC_KNEE_IP_WIDTH 6 /* DRC_KNEE_IP - [10:5] */ +#define WM8904_DRC_KNEE_OP_MASK 0x001F /* DRC_KNEE_OP - [4:0] */ +#define WM8904_DRC_KNEE_OP_SHIFT 0 /* DRC_KNEE_OP - [4:0] */ +#define WM8904_DRC_KNEE_OP_WIDTH 5 /* DRC_KNEE_OP - [4:0] */ + +/* + * R44 (0x2C) - Analogue Left Input 0 + */ +#define WM8904_LINMUTE 0x0080 /* LINMUTE */ +#define WM8904_LINMUTE_MASK 0x0080 /* LINMUTE */ +#define WM8904_LINMUTE_SHIFT 7 /* LINMUTE */ +#define WM8904_LINMUTE_WIDTH 1 /* LINMUTE */ +#define WM8904_LIN_VOL_MASK 0x001F /* LIN_VOL - [4:0] */ +#define WM8904_LIN_VOL_SHIFT 0 /* LIN_VOL - [4:0] */ +#define WM8904_LIN_VOL_WIDTH 5 /* LIN_VOL - [4:0] */ + +/* + * R45 (0x2D) - Analogue Right Input 0 + */ +#define WM8904_RINMUTE 0x0080 /* RINMUTE */ +#define WM8904_RINMUTE_MASK 0x0080 /* RINMUTE */ +#define WM8904_RINMUTE_SHIFT 7 /* RINMUTE */ +#define WM8904_RINMUTE_WIDTH 1 /* RINMUTE */ +#define WM8904_RIN_VOL_MASK 0x001F /* RIN_VOL - [4:0] */ +#define WM8904_RIN_VOL_SHIFT 0 /* RIN_VOL - [4:0] */ +#define WM8904_RIN_VOL_WIDTH 5 /* RIN_VOL - [4:0] */ + +/* + * R46 (0x2E) - Analogue Left Input 1 + */ +#define WM8904_INL_CM_ENA 0x0040 /* INL_CM_ENA */ +#define WM8904_INL_CM_ENA_MASK 0x0040 /* INL_CM_ENA */ +#define WM8904_INL_CM_ENA_SHIFT 6 /* INL_CM_ENA */ +#define WM8904_INL_CM_ENA_WIDTH 1 /* INL_CM_ENA */ +#define WM8904_L_IP_SEL_N_MASK 0x0030 /* L_IP_SEL_N - [5:4] */ +#define WM8904_L_IP_SEL_N_SHIFT 4 /* L_IP_SEL_N - [5:4] */ +#define WM8904_L_IP_SEL_N_WIDTH 2 /* L_IP_SEL_N - [5:4] */ +#define WM8904_L_IP_SEL_P_MASK 0x000C /* L_IP_SEL_P - [3:2] */ +#define WM8904_L_IP_SEL_P_SHIFT 2 /* L_IP_SEL_P - [3:2] */ +#define WM8904_L_IP_SEL_P_WIDTH 2 /* L_IP_SEL_P - [3:2] */ +#define WM8904_L_MODE_MASK 0x0003 /* L_MODE - [1:0] */ +#define WM8904_L_MODE_SHIFT 0 /* L_MODE - [1:0] */ +#define WM8904_L_MODE_WIDTH 2 /* L_MODE - [1:0] */ + +/* + * R47 (0x2F) - Analogue Right Input 1 + */ +#define WM8904_INR_CM_ENA 0x0040 /* INR_CM_ENA */ +#define WM8904_INR_CM_ENA_MASK 0x0040 /* INR_CM_ENA */ +#define WM8904_INR_CM_ENA_SHIFT 6 /* INR_CM_ENA */ +#define WM8904_INR_CM_ENA_WIDTH 1 /* INR_CM_ENA */ +#define WM8904_R_IP_SEL_N_MASK 0x0030 /* R_IP_SEL_N - [5:4] */ +#define WM8904_R_IP_SEL_N_SHIFT 4 /* R_IP_SEL_N - [5:4] */ +#define WM8904_R_IP_SEL_N_WIDTH 2 /* R_IP_SEL_N - [5:4] */ +#define WM8904_R_IP_SEL_P_MASK 0x000C /* R_IP_SEL_P - [3:2] */ +#define WM8904_R_IP_SEL_P_SHIFT 2 /* R_IP_SEL_P - [3:2] */ +#define WM8904_R_IP_SEL_P_WIDTH 2 /* R_IP_SEL_P - [3:2] */ +#define WM8904_R_MODE_MASK 0x0003 /* R_MODE - [1:0] */ +#define WM8904_R_MODE_SHIFT 0 /* R_MODE - [1:0] */ +#define WM8904_R_MODE_WIDTH 2 /* R_MODE - [1:0] */ + +/* + * R57 (0x39) - Analogue OUT1 Left + */ +#define WM8904_HPOUTL_MUTE 0x0100 /* HPOUTL_MUTE */ +#define WM8904_HPOUTL_MUTE_MASK 0x0100 /* HPOUTL_MUTE */ +#define WM8904_HPOUTL_MUTE_SHIFT 8 /* HPOUTL_MUTE */ +#define WM8904_HPOUTL_MUTE_WIDTH 1 /* HPOUTL_MUTE */ +#define WM8904_HPOUT_VU 0x0080 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_MASK 0x0080 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_SHIFT 7 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_WIDTH 1 /* HPOUT_VU */ +#define WM8904_HPOUTLZC 0x0040 /* HPOUTLZC */ +#define WM8904_HPOUTLZC_MASK 0x0040 /* HPOUTLZC */ +#define WM8904_HPOUTLZC_SHIFT 6 /* HPOUTLZC */ +#define WM8904_HPOUTLZC_WIDTH 1 /* HPOUTLZC */ +#define WM8904_HPOUTL_VOL_MASK 0x003F /* HPOUTL_VOL - [5:0] */ +#define WM8904_HPOUTL_VOL_SHIFT 0 /* HPOUTL_VOL - [5:0] */ +#define WM8904_HPOUTL_VOL_WIDTH 6 /* HPOUTL_VOL - [5:0] */ + +/* + * R58 (0x3A) - Analogue OUT1 Right + */ +#define WM8904_HPOUTR_MUTE 0x0100 /* HPOUTR_MUTE */ +#define WM8904_HPOUTR_MUTE_MASK 0x0100 /* HPOUTR_MUTE */ +#define WM8904_HPOUTR_MUTE_SHIFT 8 /* HPOUTR_MUTE */ +#define WM8904_HPOUTR_MUTE_WIDTH 1 /* HPOUTR_MUTE */ +#define WM8904_HPOUT_VU 0x0080 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_MASK 0x0080 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_SHIFT 7 /* HPOUT_VU */ +#define WM8904_HPOUT_VU_WIDTH 1 /* HPOUT_VU */ +#define WM8904_HPOUTRZC 0x0040 /* HPOUTRZC */ +#define WM8904_HPOUTRZC_MASK 0x0040 /* HPOUTRZC */ +#define WM8904_HPOUTRZC_SHIFT 6 /* HPOUTRZC */ +#define WM8904_HPOUTRZC_WIDTH 1 /* HPOUTRZC */ +#define WM8904_HPOUTR_VOL_MASK 0x003F /* HPOUTR_VOL - [5:0] */ +#define WM8904_HPOUTR_VOL_SHIFT 0 /* HPOUTR_VOL - [5:0] */ +#define WM8904_HPOUTR_VOL_WIDTH 6 /* HPOUTR_VOL - [5:0] */ + +/* + * R59 (0x3B) - Analogue OUT2 Left + */ +#define WM8904_LINEOUTL_MUTE 0x0100 /* LINEOUTL_MUTE */ +#define WM8904_LINEOUTL_MUTE_MASK 0x0100 /* LINEOUTL_MUTE */ +#define WM8904_LINEOUTL_MUTE_SHIFT 8 /* LINEOUTL_MUTE */ +#define WM8904_LINEOUTL_MUTE_WIDTH 1 /* LINEOUTL_MUTE */ +#define WM8904_LINEOUT_VU 0x0080 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_MASK 0x0080 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_SHIFT 7 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_WIDTH 1 /* LINEOUT_VU */ +#define WM8904_LINEOUTLZC 0x0040 /* LINEOUTLZC */ +#define WM8904_LINEOUTLZC_MASK 0x0040 /* LINEOUTLZC */ +#define WM8904_LINEOUTLZC_SHIFT 6 /* LINEOUTLZC */ +#define WM8904_LINEOUTLZC_WIDTH 1 /* LINEOUTLZC */ +#define WM8904_LINEOUTL_VOL_MASK 0x003F /* LINEOUTL_VOL - [5:0] */ +#define WM8904_LINEOUTL_VOL_SHIFT 0 /* LINEOUTL_VOL - [5:0] */ +#define WM8904_LINEOUTL_VOL_WIDTH 6 /* LINEOUTL_VOL - [5:0] */ + +/* + * R60 (0x3C) - Analogue OUT2 Right + */ +#define WM8904_LINEOUTR_MUTE 0x0100 /* LINEOUTR_MUTE */ +#define WM8904_LINEOUTR_MUTE_MASK 0x0100 /* LINEOUTR_MUTE */ +#define WM8904_LINEOUTR_MUTE_SHIFT 8 /* LINEOUTR_MUTE */ +#define WM8904_LINEOUTR_MUTE_WIDTH 1 /* LINEOUTR_MUTE */ +#define WM8904_LINEOUT_VU 0x0080 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_MASK 0x0080 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_SHIFT 7 /* LINEOUT_VU */ +#define WM8904_LINEOUT_VU_WIDTH 1 /* LINEOUT_VU */ +#define WM8904_LINEOUTRZC 0x0040 /* LINEOUTRZC */ +#define WM8904_LINEOUTRZC_MASK 0x0040 /* LINEOUTRZC */ +#define WM8904_LINEOUTRZC_SHIFT 6 /* LINEOUTRZC */ +#define WM8904_LINEOUTRZC_WIDTH 1 /* LINEOUTRZC */ +#define WM8904_LINEOUTR_VOL_MASK 0x003F /* LINEOUTR_VOL - [5:0] */ +#define WM8904_LINEOUTR_VOL_SHIFT 0 /* LINEOUTR_VOL - [5:0] */ +#define WM8904_LINEOUTR_VOL_WIDTH 6 /* LINEOUTR_VOL - [5:0] */ + +/* + * R61 (0x3D) - Analogue OUT12 ZC + */ +#define WM8904_HPL_BYP_ENA 0x0008 /* HPL_BYP_ENA */ +#define WM8904_HPL_BYP_ENA_MASK 0x0008 /* HPL_BYP_ENA */ +#define WM8904_HPL_BYP_ENA_SHIFT 3 /* HPL_BYP_ENA */ +#define WM8904_HPL_BYP_ENA_WIDTH 1 /* HPL_BYP_ENA */ +#define WM8904_HPR_BYP_ENA 0x0004 /* HPR_BYP_ENA */ +#define WM8904_HPR_BYP_ENA_MASK 0x0004 /* HPR_BYP_ENA */ +#define WM8904_HPR_BYP_ENA_SHIFT 2 /* HPR_BYP_ENA */ +#define WM8904_HPR_BYP_ENA_WIDTH 1 /* HPR_BYP_ENA */ +#define WM8904_LINEOUTL_BYP_ENA 0x0002 /* LINEOUTL_BYP_ENA */ +#define WM8904_LINEOUTL_BYP_ENA_MASK 0x0002 /* LINEOUTL_BYP_ENA */ +#define WM8904_LINEOUTL_BYP_ENA_SHIFT 1 /* LINEOUTL_BYP_ENA */ +#define WM8904_LINEOUTL_BYP_ENA_WIDTH 1 /* LINEOUTL_BYP_ENA */ +#define WM8904_LINEOUTR_BYP_ENA 0x0001 /* LINEOUTR_BYP_ENA */ +#define WM8904_LINEOUTR_BYP_ENA_MASK 0x0001 /* LINEOUTR_BYP_ENA */ +#define WM8904_LINEOUTR_BYP_ENA_SHIFT 0 /* LINEOUTR_BYP_ENA */ +#define WM8904_LINEOUTR_BYP_ENA_WIDTH 1 /* LINEOUTR_BYP_ENA */ + +/* + * R67 (0x43) - DC Servo 0 + */ +#define WM8904_DCS_ENA_CHAN_3 0x0008 /* DCS_ENA_CHAN_3 */ +#define WM8904_DCS_ENA_CHAN_3_MASK 0x0008 /* DCS_ENA_CHAN_3 */ +#define WM8904_DCS_ENA_CHAN_3_SHIFT 3 /* DCS_ENA_CHAN_3 */ +#define WM8904_DCS_ENA_CHAN_3_WIDTH 1 /* DCS_ENA_CHAN_3 */ +#define WM8904_DCS_ENA_CHAN_2 0x0004 /* DCS_ENA_CHAN_2 */ +#define WM8904_DCS_ENA_CHAN_2_MASK 0x0004 /* DCS_ENA_CHAN_2 */ +#define WM8904_DCS_ENA_CHAN_2_SHIFT 2 /* DCS_ENA_CHAN_2 */ +#define WM8904_DCS_ENA_CHAN_2_WIDTH 1 /* DCS_ENA_CHAN_2 */ +#define WM8904_DCS_ENA_CHAN_1 0x0002 /* DCS_ENA_CHAN_1 */ +#define WM8904_DCS_ENA_CHAN_1_MASK 0x0002 /* DCS_ENA_CHAN_1 */ +#define WM8904_DCS_ENA_CHAN_1_SHIFT 1 /* DCS_ENA_CHAN_1 */ +#define WM8904_DCS_ENA_CHAN_1_WIDTH 1 /* DCS_ENA_CHAN_1 */ +#define WM8904_DCS_ENA_CHAN_0 0x0001 /* DCS_ENA_CHAN_0 */ +#define WM8904_DCS_ENA_CHAN_0_MASK 0x0001 /* DCS_ENA_CHAN_0 */ +#define WM8904_DCS_ENA_CHAN_0_SHIFT 0 /* DCS_ENA_CHAN_0 */ +#define WM8904_DCS_ENA_CHAN_0_WIDTH 1 /* DCS_ENA_CHAN_0 */ + +/* + * R68 (0x44) - DC Servo 1 + */ +#define WM8904_DCS_TRIG_SINGLE_3 0x8000 /* DCS_TRIG_SINGLE_3 */ +#define WM8904_DCS_TRIG_SINGLE_3_MASK 0x8000 /* DCS_TRIG_SINGLE_3 */ +#define WM8904_DCS_TRIG_SINGLE_3_SHIFT 15 /* DCS_TRIG_SINGLE_3 */ +#define WM8904_DCS_TRIG_SINGLE_3_WIDTH 1 /* DCS_TRIG_SINGLE_3 */ +#define WM8904_DCS_TRIG_SINGLE_2 0x4000 /* DCS_TRIG_SINGLE_2 */ +#define WM8904_DCS_TRIG_SINGLE_2_MASK 0x4000 /* DCS_TRIG_SINGLE_2 */ +#define WM8904_DCS_TRIG_SINGLE_2_SHIFT 14 /* DCS_TRIG_SINGLE_2 */ +#define WM8904_DCS_TRIG_SINGLE_2_WIDTH 1 /* DCS_TRIG_SINGLE_2 */ +#define WM8904_DCS_TRIG_SINGLE_1 0x2000 /* DCS_TRIG_SINGLE_1 */ +#define WM8904_DCS_TRIG_SINGLE_1_MASK 0x2000 /* DCS_TRIG_SINGLE_1 */ +#define WM8904_DCS_TRIG_SINGLE_1_SHIFT 13 /* DCS_TRIG_SINGLE_1 */ +#define WM8904_DCS_TRIG_SINGLE_1_WIDTH 1 /* DCS_TRIG_SINGLE_1 */ +#define WM8904_DCS_TRIG_SINGLE_0 0x1000 /* DCS_TRIG_SINGLE_0 */ +#define WM8904_DCS_TRIG_SINGLE_0_MASK 0x1000 /* DCS_TRIG_SINGLE_0 */ +#define WM8904_DCS_TRIG_SINGLE_0_SHIFT 12 /* DCS_TRIG_SINGLE_0 */ +#define WM8904_DCS_TRIG_SINGLE_0_WIDTH 1 /* DCS_TRIG_SINGLE_0 */ +#define WM8904_DCS_TRIG_SERIES_3 0x0800 /* DCS_TRIG_SERIES_3 */ +#define WM8904_DCS_TRIG_SERIES_3_MASK 0x0800 /* DCS_TRIG_SERIES_3 */ +#define WM8904_DCS_TRIG_SERIES_3_SHIFT 11 /* DCS_TRIG_SERIES_3 */ +#define WM8904_DCS_TRIG_SERIES_3_WIDTH 1 /* DCS_TRIG_SERIES_3 */ +#define WM8904_DCS_TRIG_SERIES_2 0x0400 /* DCS_TRIG_SERIES_2 */ +#define WM8904_DCS_TRIG_SERIES_2_MASK 0x0400 /* DCS_TRIG_SERIES_2 */ +#define WM8904_DCS_TRIG_SERIES_2_SHIFT 10 /* DCS_TRIG_SERIES_2 */ +#define WM8904_DCS_TRIG_SERIES_2_WIDTH 1 /* DCS_TRIG_SERIES_2 */ +#define WM8904_DCS_TRIG_SERIES_1 0x0200 /* DCS_TRIG_SERIES_1 */ +#define WM8904_DCS_TRIG_SERIES_1_MASK 0x0200 /* DCS_TRIG_SERIES_1 */ +#define WM8904_DCS_TRIG_SERIES_1_SHIFT 9 /* DCS_TRIG_SERIES_1 */ +#define WM8904_DCS_TRIG_SERIES_1_WIDTH 1 /* DCS_TRIG_SERIES_1 */ +#define WM8904_DCS_TRIG_SERIES_0 0x0100 /* DCS_TRIG_SERIES_0 */ +#define WM8904_DCS_TRIG_SERIES_0_MASK 0x0100 /* DCS_TRIG_SERIES_0 */ +#define WM8904_DCS_TRIG_SERIES_0_SHIFT 8 /* DCS_TRIG_SERIES_0 */ +#define WM8904_DCS_TRIG_SERIES_0_WIDTH 1 /* DCS_TRIG_SERIES_0 */ +#define WM8904_DCS_TRIG_STARTUP_3 0x0080 /* DCS_TRIG_STARTUP_3 */ +#define WM8904_DCS_TRIG_STARTUP_3_MASK 0x0080 /* DCS_TRIG_STARTUP_3 */ +#define WM8904_DCS_TRIG_STARTUP_3_SHIFT 7 /* DCS_TRIG_STARTUP_3 */ +#define WM8904_DCS_TRIG_STARTUP_3_WIDTH 1 /* DCS_TRIG_STARTUP_3 */ +#define WM8904_DCS_TRIG_STARTUP_2 0x0040 /* DCS_TRIG_STARTUP_2 */ +#define WM8904_DCS_TRIG_STARTUP_2_MASK 0x0040 /* DCS_TRIG_STARTUP_2 */ +#define WM8904_DCS_TRIG_STARTUP_2_SHIFT 6 /* DCS_TRIG_STARTUP_2 */ +#define WM8904_DCS_TRIG_STARTUP_2_WIDTH 1 /* DCS_TRIG_STARTUP_2 */ +#define WM8904_DCS_TRIG_STARTUP_1 0x0020 /* DCS_TRIG_STARTUP_1 */ +#define WM8904_DCS_TRIG_STARTUP_1_MASK 0x0020 /* DCS_TRIG_STARTUP_1 */ +#define WM8904_DCS_TRIG_STARTUP_1_SHIFT 5 /* DCS_TRIG_STARTUP_1 */ +#define WM8904_DCS_TRIG_STARTUP_1_WIDTH 1 /* DCS_TRIG_STARTUP_1 */ +#define WM8904_DCS_TRIG_STARTUP_0 0x0010 /* DCS_TRIG_STARTUP_0 */ +#define WM8904_DCS_TRIG_STARTUP_0_MASK 0x0010 /* DCS_TRIG_STARTUP_0 */ +#define WM8904_DCS_TRIG_STARTUP_0_SHIFT 4 /* DCS_TRIG_STARTUP_0 */ +#define WM8904_DCS_TRIG_STARTUP_0_WIDTH 1 /* DCS_TRIG_STARTUP_0 */ +#define WM8904_DCS_TRIG_DAC_WR_3 0x0008 /* DCS_TRIG_DAC_WR_3 */ +#define WM8904_DCS_TRIG_DAC_WR_3_MASK 0x0008 /* DCS_TRIG_DAC_WR_3 */ +#define WM8904_DCS_TRIG_DAC_WR_3_SHIFT 3 /* DCS_TRIG_DAC_WR_3 */ +#define WM8904_DCS_TRIG_DAC_WR_3_WIDTH 1 /* DCS_TRIG_DAC_WR_3 */ +#define WM8904_DCS_TRIG_DAC_WR_2 0x0004 /* DCS_TRIG_DAC_WR_2 */ +#define WM8904_DCS_TRIG_DAC_WR_2_MASK 0x0004 /* DCS_TRIG_DAC_WR_2 */ +#define WM8904_DCS_TRIG_DAC_WR_2_SHIFT 2 /* DCS_TRIG_DAC_WR_2 */ +#define WM8904_DCS_TRIG_DAC_WR_2_WIDTH 1 /* DCS_TRIG_DAC_WR_2 */ +#define WM8904_DCS_TRIG_DAC_WR_1 0x0002 /* DCS_TRIG_DAC_WR_1 */ +#define WM8904_DCS_TRIG_DAC_WR_1_MASK 0x0002 /* DCS_TRIG_DAC_WR_1 */ +#define WM8904_DCS_TRIG_DAC_WR_1_SHIFT 1 /* DCS_TRIG_DAC_WR_1 */ +#define WM8904_DCS_TRIG_DAC_WR_1_WIDTH 1 /* DCS_TRIG_DAC_WR_1 */ +#define WM8904_DCS_TRIG_DAC_WR_0 0x0001 /* DCS_TRIG_DAC_WR_0 */ +#define WM8904_DCS_TRIG_DAC_WR_0_MASK 0x0001 /* DCS_TRIG_DAC_WR_0 */ +#define WM8904_DCS_TRIG_DAC_WR_0_SHIFT 0 /* DCS_TRIG_DAC_WR_0 */ +#define WM8904_DCS_TRIG_DAC_WR_0_WIDTH 1 /* DCS_TRIG_DAC_WR_0 */ + +/* + * R69 (0x45) - DC Servo 2 + */ +#define WM8904_DCS_TIMER_PERIOD_23_MASK 0x0F00 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8904_DCS_TIMER_PERIOD_23_SHIFT 8 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8904_DCS_TIMER_PERIOD_23_WIDTH 4 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8904_DCS_TIMER_PERIOD_01_MASK 0x000F /* DCS_TIMER_PERIOD_01 - [3:0] */ +#define WM8904_DCS_TIMER_PERIOD_01_SHIFT 0 /* DCS_TIMER_PERIOD_01 - [3:0] */ +#define WM8904_DCS_TIMER_PERIOD_01_WIDTH 4 /* DCS_TIMER_PERIOD_01 - [3:0] */ + +/* + * R71 (0x47) - DC Servo 4 + */ +#define WM8904_DCS_SERIES_NO_23_MASK 0x007F /* DCS_SERIES_NO_23 - [6:0] */ +#define WM8904_DCS_SERIES_NO_23_SHIFT 0 /* DCS_SERIES_NO_23 - [6:0] */ +#define WM8904_DCS_SERIES_NO_23_WIDTH 7 /* DCS_SERIES_NO_23 - [6:0] */ + +/* + * R72 (0x48) - DC Servo 5 + */ +#define WM8904_DCS_SERIES_NO_01_MASK 0x007F /* DCS_SERIES_NO_01 - [6:0] */ +#define WM8904_DCS_SERIES_NO_01_SHIFT 0 /* DCS_SERIES_NO_01 - [6:0] */ +#define WM8904_DCS_SERIES_NO_01_WIDTH 7 /* DCS_SERIES_NO_01 - [6:0] */ + +/* + * R73 (0x49) - DC Servo 6 + */ +#define WM8904_DCS_DAC_WR_VAL_3_MASK 0x00FF /* DCS_DAC_WR_VAL_3 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_3_SHIFT 0 /* DCS_DAC_WR_VAL_3 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_3_WIDTH 8 /* DCS_DAC_WR_VAL_3 - [7:0] */ + +/* + * R74 (0x4A) - DC Servo 7 + */ +#define WM8904_DCS_DAC_WR_VAL_2_MASK 0x00FF /* DCS_DAC_WR_VAL_2 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_2_SHIFT 0 /* DCS_DAC_WR_VAL_2 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_2_WIDTH 8 /* DCS_DAC_WR_VAL_2 - [7:0] */ + +/* + * R75 (0x4B) - DC Servo 8 + */ +#define WM8904_DCS_DAC_WR_VAL_1_MASK 0x00FF /* DCS_DAC_WR_VAL_1 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_1_SHIFT 0 /* DCS_DAC_WR_VAL_1 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_1_WIDTH 8 /* DCS_DAC_WR_VAL_1 - [7:0] */ + +/* + * R76 (0x4C) - DC Servo 9 + */ +#define WM8904_DCS_DAC_WR_VAL_0_MASK 0x00FF /* DCS_DAC_WR_VAL_0 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_0_SHIFT 0 /* DCS_DAC_WR_VAL_0 - [7:0] */ +#define WM8904_DCS_DAC_WR_VAL_0_WIDTH 8 /* DCS_DAC_WR_VAL_0 - [7:0] */ + +/* + * R77 (0x4D) - DC Servo Readback 0 + */ +#define WM8904_DCS_CAL_COMPLETE_MASK 0x0F00 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8904_DCS_CAL_COMPLETE_SHIFT 8 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8904_DCS_CAL_COMPLETE_WIDTH 4 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8904_DCS_DAC_WR_COMPLETE_MASK 0x00F0 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8904_DCS_DAC_WR_COMPLETE_SHIFT 4 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8904_DCS_DAC_WR_COMPLETE_WIDTH 4 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8904_DCS_STARTUP_COMPLETE_MASK 0x000F /* DCS_STARTUP_COMPLETE - [3:0] */ +#define WM8904_DCS_STARTUP_COMPLETE_SHIFT 0 /* DCS_STARTUP_COMPLETE - [3:0] */ +#define WM8904_DCS_STARTUP_COMPLETE_WIDTH 4 /* DCS_STARTUP_COMPLETE - [3:0] */ + +/* + * R90 (0x5A) - Analogue HP 0 + */ +#define WM8904_HPL_RMV_SHORT 0x0080 /* HPL_RMV_SHORT */ +#define WM8904_HPL_RMV_SHORT_MASK 0x0080 /* HPL_RMV_SHORT */ +#define WM8904_HPL_RMV_SHORT_SHIFT 7 /* HPL_RMV_SHORT */ +#define WM8904_HPL_RMV_SHORT_WIDTH 1 /* HPL_RMV_SHORT */ +#define WM8904_HPL_ENA_OUTP 0x0040 /* HPL_ENA_OUTP */ +#define WM8904_HPL_ENA_OUTP_MASK 0x0040 /* HPL_ENA_OUTP */ +#define WM8904_HPL_ENA_OUTP_SHIFT 6 /* HPL_ENA_OUTP */ +#define WM8904_HPL_ENA_OUTP_WIDTH 1 /* HPL_ENA_OUTP */ +#define WM8904_HPL_ENA_DLY 0x0020 /* HPL_ENA_DLY */ +#define WM8904_HPL_ENA_DLY_MASK 0x0020 /* HPL_ENA_DLY */ +#define WM8904_HPL_ENA_DLY_SHIFT 5 /* HPL_ENA_DLY */ +#define WM8904_HPL_ENA_DLY_WIDTH 1 /* HPL_ENA_DLY */ +#define WM8904_HPL_ENA 0x0010 /* HPL_ENA */ +#define WM8904_HPL_ENA_MASK 0x0010 /* HPL_ENA */ +#define WM8904_HPL_ENA_SHIFT 4 /* HPL_ENA */ +#define WM8904_HPL_ENA_WIDTH 1 /* HPL_ENA */ +#define WM8904_HPR_RMV_SHORT 0x0008 /* HPR_RMV_SHORT */ +#define WM8904_HPR_RMV_SHORT_MASK 0x0008 /* HPR_RMV_SHORT */ +#define WM8904_HPR_RMV_SHORT_SHIFT 3 /* HPR_RMV_SHORT */ +#define WM8904_HPR_RMV_SHORT_WIDTH 1 /* HPR_RMV_SHORT */ +#define WM8904_HPR_ENA_OUTP 0x0004 /* HPR_ENA_OUTP */ +#define WM8904_HPR_ENA_OUTP_MASK 0x0004 /* HPR_ENA_OUTP */ +#define WM8904_HPR_ENA_OUTP_SHIFT 2 /* HPR_ENA_OUTP */ +#define WM8904_HPR_ENA_OUTP_WIDTH 1 /* HPR_ENA_OUTP */ +#define WM8904_HPR_ENA_DLY 0x0002 /* HPR_ENA_DLY */ +#define WM8904_HPR_ENA_DLY_MASK 0x0002 /* HPR_ENA_DLY */ +#define WM8904_HPR_ENA_DLY_SHIFT 1 /* HPR_ENA_DLY */ +#define WM8904_HPR_ENA_DLY_WIDTH 1 /* HPR_ENA_DLY */ +#define WM8904_HPR_ENA 0x0001 /* HPR_ENA */ +#define WM8904_HPR_ENA_MASK 0x0001 /* HPR_ENA */ +#define WM8904_HPR_ENA_SHIFT 0 /* HPR_ENA */ +#define WM8904_HPR_ENA_WIDTH 1 /* HPR_ENA */ + +/* + * R94 (0x5E) - Analogue Lineout 0 + */ +#define WM8904_LINEOUTL_RMV_SHORT 0x0080 /* LINEOUTL_RMV_SHORT */ +#define WM8904_LINEOUTL_RMV_SHORT_MASK 0x0080 /* LINEOUTL_RMV_SHORT */ +#define WM8904_LINEOUTL_RMV_SHORT_SHIFT 7 /* LINEOUTL_RMV_SHORT */ +#define WM8904_LINEOUTL_RMV_SHORT_WIDTH 1 /* LINEOUTL_RMV_SHORT */ +#define WM8904_LINEOUTL_ENA_OUTP 0x0040 /* LINEOUTL_ENA_OUTP */ +#define WM8904_LINEOUTL_ENA_OUTP_MASK 0x0040 /* LINEOUTL_ENA_OUTP */ +#define WM8904_LINEOUTL_ENA_OUTP_SHIFT 6 /* LINEOUTL_ENA_OUTP */ +#define WM8904_LINEOUTL_ENA_OUTP_WIDTH 1 /* LINEOUTL_ENA_OUTP */ +#define WM8904_LINEOUTL_ENA_DLY 0x0020 /* LINEOUTL_ENA_DLY */ +#define WM8904_LINEOUTL_ENA_DLY_MASK 0x0020 /* LINEOUTL_ENA_DLY */ +#define WM8904_LINEOUTL_ENA_DLY_SHIFT 5 /* LINEOUTL_ENA_DLY */ +#define WM8904_LINEOUTL_ENA_DLY_WIDTH 1 /* LINEOUTL_ENA_DLY */ +#define WM8904_LINEOUTL_ENA 0x0010 /* LINEOUTL_ENA */ +#define WM8904_LINEOUTL_ENA_MASK 0x0010 /* LINEOUTL_ENA */ +#define WM8904_LINEOUTL_ENA_SHIFT 4 /* LINEOUTL_ENA */ +#define WM8904_LINEOUTL_ENA_WIDTH 1 /* LINEOUTL_ENA */ +#define WM8904_LINEOUTR_RMV_SHORT 0x0008 /* LINEOUTR_RMV_SHORT */ +#define WM8904_LINEOUTR_RMV_SHORT_MASK 0x0008 /* LINEOUTR_RMV_SHORT */ +#define WM8904_LINEOUTR_RMV_SHORT_SHIFT 3 /* LINEOUTR_RMV_SHORT */ +#define WM8904_LINEOUTR_RMV_SHORT_WIDTH 1 /* LINEOUTR_RMV_SHORT */ +#define WM8904_LINEOUTR_ENA_OUTP 0x0004 /* LINEOUTR_ENA_OUTP */ +#define WM8904_LINEOUTR_ENA_OUTP_MASK 0x0004 /* LINEOUTR_ENA_OUTP */ +#define WM8904_LINEOUTR_ENA_OUTP_SHIFT 2 /* LINEOUTR_ENA_OUTP */ +#define WM8904_LINEOUTR_ENA_OUTP_WIDTH 1 /* LINEOUTR_ENA_OUTP */ +#define WM8904_LINEOUTR_ENA_DLY 0x0002 /* LINEOUTR_ENA_DLY */ +#define WM8904_LINEOUTR_ENA_DLY_MASK 0x0002 /* LINEOUTR_ENA_DLY */ +#define WM8904_LINEOUTR_ENA_DLY_SHIFT 1 /* LINEOUTR_ENA_DLY */ +#define WM8904_LINEOUTR_ENA_DLY_WIDTH 1 /* LINEOUTR_ENA_DLY */ +#define WM8904_LINEOUTR_ENA 0x0001 /* LINEOUTR_ENA */ +#define WM8904_LINEOUTR_ENA_MASK 0x0001 /* LINEOUTR_ENA */ +#define WM8904_LINEOUTR_ENA_SHIFT 0 /* LINEOUTR_ENA */ +#define WM8904_LINEOUTR_ENA_WIDTH 1 /* LINEOUTR_ENA */ + +/* + * R98 (0x62) - Charge Pump 0 + */ +#define WM8904_CP_ENA 0x0001 /* CP_ENA */ +#define WM8904_CP_ENA_MASK 0x0001 /* CP_ENA */ +#define WM8904_CP_ENA_SHIFT 0 /* CP_ENA */ +#define WM8904_CP_ENA_WIDTH 1 /* CP_ENA */ + +/* + * R104 (0x68) - Class W 0 + */ +#define WM8904_CP_DYN_PWR 0x0001 /* CP_DYN_PWR */ +#define WM8904_CP_DYN_PWR_MASK 0x0001 /* CP_DYN_PWR */ +#define WM8904_CP_DYN_PWR_SHIFT 0 /* CP_DYN_PWR */ +#define WM8904_CP_DYN_PWR_WIDTH 1 /* CP_DYN_PWR */ + +/* + * R108 (0x6C) - Write Sequencer 0 + */ +#define WM8904_WSEQ_ENA 0x0100 /* WSEQ_ENA */ +#define WM8904_WSEQ_ENA_MASK 0x0100 /* WSEQ_ENA */ +#define WM8904_WSEQ_ENA_SHIFT 8 /* WSEQ_ENA */ +#define WM8904_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */ +#define WM8904_WSEQ_WRITE_INDEX_MASK 0x001F /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8904_WSEQ_WRITE_INDEX_SHIFT 0 /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8904_WSEQ_WRITE_INDEX_WIDTH 5 /* WSEQ_WRITE_INDEX - [4:0] */ + +/* + * R109 (0x6D) - Write Sequencer 1 + */ +#define WM8904_WSEQ_DATA_WIDTH_MASK 0x7000 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8904_WSEQ_DATA_WIDTH_SHIFT 12 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8904_WSEQ_DATA_WIDTH_WIDTH 3 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8904_WSEQ_DATA_START_MASK 0x0F00 /* WSEQ_DATA_START - [11:8] */ +#define WM8904_WSEQ_DATA_START_SHIFT 8 /* WSEQ_DATA_START - [11:8] */ +#define WM8904_WSEQ_DATA_START_WIDTH 4 /* WSEQ_DATA_START - [11:8] */ +#define WM8904_WSEQ_ADDR_MASK 0x00FF /* WSEQ_ADDR - [7:0] */ +#define WM8904_WSEQ_ADDR_SHIFT 0 /* WSEQ_ADDR - [7:0] */ +#define WM8904_WSEQ_ADDR_WIDTH 8 /* WSEQ_ADDR - [7:0] */ + +/* + * R110 (0x6E) - Write Sequencer 2 + */ +#define WM8904_WSEQ_EOS 0x4000 /* WSEQ_EOS */ +#define WM8904_WSEQ_EOS_MASK 0x4000 /* WSEQ_EOS */ +#define WM8904_WSEQ_EOS_SHIFT 14 /* WSEQ_EOS */ +#define WM8904_WSEQ_EOS_WIDTH 1 /* WSEQ_EOS */ +#define WM8904_WSEQ_DELAY_MASK 0x0F00 /* WSEQ_DELAY - [11:8] */ +#define WM8904_WSEQ_DELAY_SHIFT 8 /* WSEQ_DELAY - [11:8] */ +#define WM8904_WSEQ_DELAY_WIDTH 4 /* WSEQ_DELAY - [11:8] */ +#define WM8904_WSEQ_DATA_MASK 0x00FF /* WSEQ_DATA - [7:0] */ +#define WM8904_WSEQ_DATA_SHIFT 0 /* WSEQ_DATA - [7:0] */ +#define WM8904_WSEQ_DATA_WIDTH 8 /* WSEQ_DATA - [7:0] */ + +/* + * R111 (0x6F) - Write Sequencer 3 + */ +#define WM8904_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */ +#define WM8904_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */ +#define WM8904_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */ +#define WM8904_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */ +#define WM8904_WSEQ_START 0x0100 /* WSEQ_START */ +#define WM8904_WSEQ_START_MASK 0x0100 /* WSEQ_START */ +#define WM8904_WSEQ_START_SHIFT 8 /* WSEQ_START */ +#define WM8904_WSEQ_START_WIDTH 1 /* WSEQ_START */ +#define WM8904_WSEQ_START_INDEX_MASK 0x003F /* WSEQ_START_INDEX - [5:0] */ +#define WM8904_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [5:0] */ +#define WM8904_WSEQ_START_INDEX_WIDTH 6 /* WSEQ_START_INDEX - [5:0] */ + +/* + * R112 (0x70) - Write Sequencer 4 + */ +#define WM8904_WSEQ_CURRENT_INDEX_MASK 0x03F0 /* WSEQ_CURRENT_INDEX - [9:4] */ +#define WM8904_WSEQ_CURRENT_INDEX_SHIFT 4 /* WSEQ_CURRENT_INDEX - [9:4] */ +#define WM8904_WSEQ_CURRENT_INDEX_WIDTH 6 /* WSEQ_CURRENT_INDEX - [9:4] */ +#define WM8904_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */ +#define WM8904_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */ +#define WM8904_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */ +#define WM8904_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */ + +/* + * R116 (0x74) - FLL Control 1 + */ +#define WM8904_FLL_FRACN_ENA 0x0004 /* FLL_FRACN_ENA */ +#define WM8904_FLL_FRACN_ENA_MASK 0x0004 /* FLL_FRACN_ENA */ +#define WM8904_FLL_FRACN_ENA_SHIFT 2 /* FLL_FRACN_ENA */ +#define WM8904_FLL_FRACN_ENA_WIDTH 1 /* FLL_FRACN_ENA */ +#define WM8904_FLL_OSC_ENA 0x0002 /* FLL_OSC_ENA */ +#define WM8904_FLL_OSC_ENA_MASK 0x0002 /* FLL_OSC_ENA */ +#define WM8904_FLL_OSC_ENA_SHIFT 1 /* FLL_OSC_ENA */ +#define WM8904_FLL_OSC_ENA_WIDTH 1 /* FLL_OSC_ENA */ +#define WM8904_FLL_ENA 0x0001 /* FLL_ENA */ +#define WM8904_FLL_ENA_MASK 0x0001 /* FLL_ENA */ +#define WM8904_FLL_ENA_SHIFT 0 /* FLL_ENA */ +#define WM8904_FLL_ENA_WIDTH 1 /* FLL_ENA */ + +/* + * R117 (0x75) - FLL Control 2 + */ +#define WM8904_FLL_OUTDIV_MASK 0x3F00 /* FLL_OUTDIV - [13:8] */ +#define WM8904_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [13:8] */ +#define WM8904_FLL_OUTDIV_WIDTH 6 /* FLL_OUTDIV - [13:8] */ +#define WM8904_FLL_CTRL_RATE_MASK 0x0070 /* FLL_CTRL_RATE - [6:4] */ +#define WM8904_FLL_CTRL_RATE_SHIFT 4 /* FLL_CTRL_RATE - [6:4] */ +#define WM8904_FLL_CTRL_RATE_WIDTH 3 /* FLL_CTRL_RATE - [6:4] */ +#define WM8904_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */ +#define WM8904_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */ +#define WM8904_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */ + +/* + * R118 (0x76) - FLL Control 3 + */ +#define WM8904_FLL_K_MASK 0xFFFF /* FLL_K - [15:0] */ +#define WM8904_FLL_K_SHIFT 0 /* FLL_K - [15:0] */ +#define WM8904_FLL_K_WIDTH 16 /* FLL_K - [15:0] */ + +/* + * R119 (0x77) - FLL Control 4 + */ +#define WM8904_FLL_N_MASK 0x7FE0 /* FLL_N - [14:5] */ +#define WM8904_FLL_N_SHIFT 5 /* FLL_N - [14:5] */ +#define WM8904_FLL_N_WIDTH 10 /* FLL_N - [14:5] */ +#define WM8904_FLL_GAIN_MASK 0x000F /* FLL_GAIN - [3:0] */ +#define WM8904_FLL_GAIN_SHIFT 0 /* FLL_GAIN - [3:0] */ +#define WM8904_FLL_GAIN_WIDTH 4 /* FLL_GAIN - [3:0] */ + +/* + * R120 (0x78) - FLL Control 5 + */ +#define WM8904_FLL_CLK_REF_DIV_MASK 0x0018 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8904_FLL_CLK_REF_DIV_SHIFT 3 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8904_FLL_CLK_REF_DIV_WIDTH 2 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8904_FLL_CLK_REF_SRC_MASK 0x0003 /* FLL_CLK_REF_SRC - [1:0] */ +#define WM8904_FLL_CLK_REF_SRC_SHIFT 0 /* FLL_CLK_REF_SRC - [1:0] */ +#define WM8904_FLL_CLK_REF_SRC_WIDTH 2 /* FLL_CLK_REF_SRC - [1:0] */ + +/* + * R121 (0x79) - GPIO Control 1 + */ +#define WM8904_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_MASK 0x0020 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_SHIFT 5 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_WIDTH 1 /* GPIO1_PU */ +#define WM8904_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_MASK 0x0010 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_SHIFT 4 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_WIDTH 1 /* GPIO1_PD */ +#define WM8904_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ +#define WM8904_GPIO1_SEL_SHIFT 0 /* GPIO1_SEL - [3:0] */ +#define WM8904_GPIO1_SEL_WIDTH 4 /* GPIO1_SEL - [3:0] */ + +/* + * R122 (0x7A) - GPIO Control 2 + */ +#define WM8904_GPIO2_PU 0x0020 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_MASK 0x0020 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_SHIFT 5 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_WIDTH 1 /* GPIO2_PU */ +#define WM8904_GPIO2_PD 0x0010 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_MASK 0x0010 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_SHIFT 4 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_WIDTH 1 /* GPIO2_PD */ +#define WM8904_GPIO2_SEL_MASK 0x000F /* GPIO2_SEL - [3:0] */ +#define WM8904_GPIO2_SEL_SHIFT 0 /* GPIO2_SEL - [3:0] */ +#define WM8904_GPIO2_SEL_WIDTH 4 /* GPIO2_SEL - [3:0] */ + +/* + * R123 (0x7B) - GPIO Control 3 + */ +#define WM8904_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_MASK 0x0020 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_SHIFT 5 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_WIDTH 1 /* GPIO3_PU */ +#define WM8904_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_MASK 0x0010 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_SHIFT 4 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_WIDTH 1 /* GPIO3_PD */ +#define WM8904_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ +#define WM8904_GPIO3_SEL_SHIFT 0 /* GPIO3_SEL - [3:0] */ +#define WM8904_GPIO3_SEL_WIDTH 4 /* GPIO3_SEL - [3:0] */ + +/* + * R124 (0x7C) - GPIO Control 4 + */ +#define WM8904_GPI7_ENA 0x0200 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_MASK 0x0200 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_SHIFT 9 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_WIDTH 1 /* GPI7_ENA */ +#define WM8904_GPI8_ENA 0x0100 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_MASK 0x0100 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_SHIFT 8 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_WIDTH 1 /* GPI8_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA 0x0080 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_MASK 0x0080 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_SHIFT 7 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_WIDTH 1 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_SEL_MASK 0x000F /* GPIO_BCLK_SEL - [3:0] */ +#define WM8904_GPIO_BCLK_SEL_SHIFT 0 /* GPIO_BCLK_SEL - [3:0] */ +#define WM8904_GPIO_BCLK_SEL_WIDTH 4 /* GPIO_BCLK_SEL - [3:0] */ + +/* + * R126 (0x7E) - Digital Pulls + */ +#define WM8904_MCLK_PU 0x0080 /* MCLK_PU */ +#define WM8904_MCLK_PU_MASK 0x0080 /* MCLK_PU */ +#define WM8904_MCLK_PU_SHIFT 7 /* MCLK_PU */ +#define WM8904_MCLK_PU_WIDTH 1 /* MCLK_PU */ +#define WM8904_MCLK_PD 0x0040 /* MCLK_PD */ +#define WM8904_MCLK_PD_MASK 0x0040 /* MCLK_PD */ +#define WM8904_MCLK_PD_SHIFT 6 /* MCLK_PD */ +#define WM8904_MCLK_PD_WIDTH 1 /* MCLK_PD */ +#define WM8904_DACDAT_PU 0x0020 /* DACDAT_PU */ +#define WM8904_DACDAT_PU_MASK 0x0020 /* DACDAT_PU */ +#define WM8904_DACDAT_PU_SHIFT 5 /* DACDAT_PU */ +#define WM8904_DACDAT_PU_WIDTH 1 /* DACDAT_PU */ +#define WM8904_DACDAT_PD 0x0010 /* DACDAT_PD */ +#define WM8904_DACDAT_PD_MASK 0x0010 /* DACDAT_PD */ +#define WM8904_DACDAT_PD_SHIFT 4 /* DACDAT_PD */ +#define WM8904_DACDAT_PD_WIDTH 1 /* DACDAT_PD */ +#define WM8904_LRCLK_PU 0x0008 /* LRCLK_PU */ +#define WM8904_LRCLK_PU_MASK 0x0008 /* LRCLK_PU */ +#define WM8904_LRCLK_PU_SHIFT 3 /* LRCLK_PU */ +#define WM8904_LRCLK_PU_WIDTH 1 /* LRCLK_PU */ +#define WM8904_LRCLK_PD 0x0004 /* LRCLK_PD */ +#define WM8904_LRCLK_PD_MASK 0x0004 /* LRCLK_PD */ +#define WM8904_LRCLK_PD_SHIFT 2 /* LRCLK_PD */ +#define WM8904_LRCLK_PD_WIDTH 1 /* LRCLK_PD */ +#define WM8904_BCLK_PU 0x0002 /* BCLK_PU */ +#define WM8904_BCLK_PU_MASK 0x0002 /* BCLK_PU */ +#define WM8904_BCLK_PU_SHIFT 1 /* BCLK_PU */ +#define WM8904_BCLK_PU_WIDTH 1 /* BCLK_PU */ +#define WM8904_BCLK_PD 0x0001 /* BCLK_PD */ +#define WM8904_BCLK_PD_MASK 0x0001 /* BCLK_PD */ +#define WM8904_BCLK_PD_SHIFT 0 /* BCLK_PD */ +#define WM8904_BCLK_PD_WIDTH 1 /* BCLK_PD */ + +/* + * R127 (0x7F) - Interrupt Status + */ +#define WM8904_IRQ 0x0400 /* IRQ */ +#define WM8904_IRQ_MASK 0x0400 /* IRQ */ +#define WM8904_IRQ_SHIFT 10 /* IRQ */ +#define WM8904_IRQ_WIDTH 1 /* IRQ */ +#define WM8904_GPIO_BCLK_EINT 0x0200 /* GPIO_BCLK_EINT */ +#define WM8904_GPIO_BCLK_EINT_MASK 0x0200 /* GPIO_BCLK_EINT */ +#define WM8904_GPIO_BCLK_EINT_SHIFT 9 /* GPIO_BCLK_EINT */ +#define WM8904_GPIO_BCLK_EINT_WIDTH 1 /* GPIO_BCLK_EINT */ +#define WM8904_WSEQ_EINT 0x0100 /* WSEQ_EINT */ +#define WM8904_WSEQ_EINT_MASK 0x0100 /* WSEQ_EINT */ +#define WM8904_WSEQ_EINT_SHIFT 8 /* WSEQ_EINT */ +#define WM8904_WSEQ_EINT_WIDTH 1 /* WSEQ_EINT */ +#define WM8904_GPIO3_EINT 0x0080 /* GPIO3_EINT */ +#define WM8904_GPIO3_EINT_MASK 0x0080 /* GPIO3_EINT */ +#define WM8904_GPIO3_EINT_SHIFT 7 /* GPIO3_EINT */ +#define WM8904_GPIO3_EINT_WIDTH 1 /* GPIO3_EINT */ +#define WM8904_GPIO2_EINT 0x0040 /* GPIO2_EINT */ +#define WM8904_GPIO2_EINT_MASK 0x0040 /* GPIO2_EINT */ +#define WM8904_GPIO2_EINT_SHIFT 6 /* GPIO2_EINT */ +#define WM8904_GPIO2_EINT_WIDTH 1 /* GPIO2_EINT */ +#define WM8904_GPIO1_EINT 0x0020 /* GPIO1_EINT */ +#define WM8904_GPIO1_EINT_MASK 0x0020 /* GPIO1_EINT */ +#define WM8904_GPIO1_EINT_SHIFT 5 /* GPIO1_EINT */ +#define WM8904_GPIO1_EINT_WIDTH 1 /* GPIO1_EINT */ +#define WM8904_GPI8_EINT 0x0010 /* GPI8_EINT */ +#define WM8904_GPI8_EINT_MASK 0x0010 /* GPI8_EINT */ +#define WM8904_GPI8_EINT_SHIFT 4 /* GPI8_EINT */ +#define WM8904_GPI8_EINT_WIDTH 1 /* GPI8_EINT */ +#define WM8904_GPI7_EINT 0x0008 /* GPI7_EINT */ +#define WM8904_GPI7_EINT_MASK 0x0008 /* GPI7_EINT */ +#define WM8904_GPI7_EINT_SHIFT 3 /* GPI7_EINT */ +#define WM8904_GPI7_EINT_WIDTH 1 /* GPI7_EINT */ +#define WM8904_FLL_LOCK_EINT 0x0004 /* FLL_LOCK_EINT */ +#define WM8904_FLL_LOCK_EINT_MASK 0x0004 /* FLL_LOCK_EINT */ +#define WM8904_FLL_LOCK_EINT_SHIFT 2 /* FLL_LOCK_EINT */ +#define WM8904_FLL_LOCK_EINT_WIDTH 1 /* FLL_LOCK_EINT */ +#define WM8904_MIC_SHRT_EINT 0x0002 /* MIC_SHRT_EINT */ +#define WM8904_MIC_SHRT_EINT_MASK 0x0002 /* MIC_SHRT_EINT */ +#define WM8904_MIC_SHRT_EINT_SHIFT 1 /* MIC_SHRT_EINT */ +#define WM8904_MIC_SHRT_EINT_WIDTH 1 /* MIC_SHRT_EINT */ +#define WM8904_MIC_DET_EINT 0x0001 /* MIC_DET_EINT */ +#define WM8904_MIC_DET_EINT_MASK 0x0001 /* MIC_DET_EINT */ +#define WM8904_MIC_DET_EINT_SHIFT 0 /* MIC_DET_EINT */ +#define WM8904_MIC_DET_EINT_WIDTH 1 /* MIC_DET_EINT */ + +/* + * R128 (0x80) - Interrupt Status Mask + */ +#define WM8904_IM_GPIO_BCLK_EINT 0x0200 /* IM_GPIO_BCLK_EINT */ +#define WM8904_IM_GPIO_BCLK_EINT_MASK 0x0200 /* IM_GPIO_BCLK_EINT */ +#define WM8904_IM_GPIO_BCLK_EINT_SHIFT 9 /* IM_GPIO_BCLK_EINT */ +#define WM8904_IM_GPIO_BCLK_EINT_WIDTH 1 /* IM_GPIO_BCLK_EINT */ +#define WM8904_IM_WSEQ_EINT 0x0100 /* IM_WSEQ_EINT */ +#define WM8904_IM_WSEQ_EINT_MASK 0x0100 /* IM_WSEQ_EINT */ +#define WM8904_IM_WSEQ_EINT_SHIFT 8 /* IM_WSEQ_EINT */ +#define WM8904_IM_WSEQ_EINT_WIDTH 1 /* IM_WSEQ_EINT */ +#define WM8904_IM_GPIO3_EINT 0x0080 /* IM_GPIO3_EINT */ +#define WM8904_IM_GPIO3_EINT_MASK 0x0080 /* IM_GPIO3_EINT */ +#define WM8904_IM_GPIO3_EINT_SHIFT 7 /* IM_GPIO3_EINT */ +#define WM8904_IM_GPIO3_EINT_WIDTH 1 /* IM_GPIO3_EINT */ +#define WM8904_IM_GPIO2_EINT 0x0040 /* IM_GPIO2_EINT */ +#define WM8904_IM_GPIO2_EINT_MASK 0x0040 /* IM_GPIO2_EINT */ +#define WM8904_IM_GPIO2_EINT_SHIFT 6 /* IM_GPIO2_EINT */ +#define WM8904_IM_GPIO2_EINT_WIDTH 1 /* IM_GPIO2_EINT */ +#define WM8904_IM_GPIO1_EINT 0x0020 /* IM_GPIO1_EINT */ +#define WM8904_IM_GPIO1_EINT_MASK 0x0020 /* IM_GPIO1_EINT */ +#define WM8904_IM_GPIO1_EINT_SHIFT 5 /* IM_GPIO1_EINT */ +#define WM8904_IM_GPIO1_EINT_WIDTH 1 /* IM_GPIO1_EINT */ +#define WM8904_IM_GPI8_EINT 0x0010 /* IM_GPI8_EINT */ +#define WM8904_IM_GPI8_EINT_MASK 0x0010 /* IM_GPI8_EINT */ +#define WM8904_IM_GPI8_EINT_SHIFT 4 /* IM_GPI8_EINT */ +#define WM8904_IM_GPI8_EINT_WIDTH 1 /* IM_GPI8_EINT */ +#define WM8904_IM_GPI7_EINT 0x0008 /* IM_GPI7_EINT */ +#define WM8904_IM_GPI7_EINT_MASK 0x0008 /* IM_GPI7_EINT */ +#define WM8904_IM_GPI7_EINT_SHIFT 3 /* IM_GPI7_EINT */ +#define WM8904_IM_GPI7_EINT_WIDTH 1 /* IM_GPI7_EINT */ +#define WM8904_IM_FLL_LOCK_EINT 0x0004 /* IM_FLL_LOCK_EINT */ +#define WM8904_IM_FLL_LOCK_EINT_MASK 0x0004 /* IM_FLL_LOCK_EINT */ +#define WM8904_IM_FLL_LOCK_EINT_SHIFT 2 /* IM_FLL_LOCK_EINT */ +#define WM8904_IM_FLL_LOCK_EINT_WIDTH 1 /* IM_FLL_LOCK_EINT */ +#define WM8904_IM_MIC_SHRT_EINT 0x0002 /* IM_MIC_SHRT_EINT */ +#define WM8904_IM_MIC_SHRT_EINT_MASK 0x0002 /* IM_MIC_SHRT_EINT */ +#define WM8904_IM_MIC_SHRT_EINT_SHIFT 1 /* IM_MIC_SHRT_EINT */ +#define WM8904_IM_MIC_SHRT_EINT_WIDTH 1 /* IM_MIC_SHRT_EINT */ +#define WM8904_IM_MIC_DET_EINT 0x0001 /* IM_MIC_DET_EINT */ +#define WM8904_IM_MIC_DET_EINT_MASK 0x0001 /* IM_MIC_DET_EINT */ +#define WM8904_IM_MIC_DET_EINT_SHIFT 0 /* IM_MIC_DET_EINT */ +#define WM8904_IM_MIC_DET_EINT_WIDTH 1 /* IM_MIC_DET_EINT */ + +/* + * R129 (0x81) - Interrupt Polarity + */ +#define WM8904_GPIO_BCLK_EINT_POL 0x0200 /* GPIO_BCLK_EINT_POL */ +#define WM8904_GPIO_BCLK_EINT_POL_MASK 0x0200 /* GPIO_BCLK_EINT_POL */ +#define WM8904_GPIO_BCLK_EINT_POL_SHIFT 9 /* GPIO_BCLK_EINT_POL */ +#define WM8904_GPIO_BCLK_EINT_POL_WIDTH 1 /* GPIO_BCLK_EINT_POL */ +#define WM8904_WSEQ_EINT_POL 0x0100 /* WSEQ_EINT_POL */ +#define WM8904_WSEQ_EINT_POL_MASK 0x0100 /* WSEQ_EINT_POL */ +#define WM8904_WSEQ_EINT_POL_SHIFT 8 /* WSEQ_EINT_POL */ +#define WM8904_WSEQ_EINT_POL_WIDTH 1 /* WSEQ_EINT_POL */ +#define WM8904_GPIO3_EINT_POL 0x0080 /* GPIO3_EINT_POL */ +#define WM8904_GPIO3_EINT_POL_MASK 0x0080 /* GPIO3_EINT_POL */ +#define WM8904_GPIO3_EINT_POL_SHIFT 7 /* GPIO3_EINT_POL */ +#define WM8904_GPIO3_EINT_POL_WIDTH 1 /* GPIO3_EINT_POL */ +#define WM8904_GPIO2_EINT_POL 0x0040 /* GPIO2_EINT_POL */ +#define WM8904_GPIO2_EINT_POL_MASK 0x0040 /* GPIO2_EINT_POL */ +#define WM8904_GPIO2_EINT_POL_SHIFT 6 /* GPIO2_EINT_POL */ +#define WM8904_GPIO2_EINT_POL_WIDTH 1 /* GPIO2_EINT_POL */ +#define WM8904_GPIO1_EINT_POL 0x0020 /* GPIO1_EINT_POL */ +#define WM8904_GPIO1_EINT_POL_MASK 0x0020 /* GPIO1_EINT_POL */ +#define WM8904_GPIO1_EINT_POL_SHIFT 5 /* GPIO1_EINT_POL */ +#define WM8904_GPIO1_EINT_POL_WIDTH 1 /* GPIO1_EINT_POL */ +#define WM8904_GPI8_EINT_POL 0x0010 /* GPI8_EINT_POL */ +#define WM8904_GPI8_EINT_POL_MASK 0x0010 /* GPI8_EINT_POL */ +#define WM8904_GPI8_EINT_POL_SHIFT 4 /* GPI8_EINT_POL */ +#define WM8904_GPI8_EINT_POL_WIDTH 1 /* GPI8_EINT_POL */ +#define WM8904_GPI7_EINT_POL 0x0008 /* GPI7_EINT_POL */ +#define WM8904_GPI7_EINT_POL_MASK 0x0008 /* GPI7_EINT_POL */ +#define WM8904_GPI7_EINT_POL_SHIFT 3 /* GPI7_EINT_POL */ +#define WM8904_GPI7_EINT_POL_WIDTH 1 /* GPI7_EINT_POL */ +#define WM8904_FLL_LOCK_EINT_POL 0x0004 /* FLL_LOCK_EINT_POL */ +#define WM8904_FLL_LOCK_EINT_POL_MASK 0x0004 /* FLL_LOCK_EINT_POL */ +#define WM8904_FLL_LOCK_EINT_POL_SHIFT 2 /* FLL_LOCK_EINT_POL */ +#define WM8904_FLL_LOCK_EINT_POL_WIDTH 1 /* FLL_LOCK_EINT_POL */ +#define WM8904_MIC_SHRT_EINT_POL 0x0002 /* MIC_SHRT_EINT_POL */ +#define WM8904_MIC_SHRT_EINT_POL_MASK 0x0002 /* MIC_SHRT_EINT_POL */ +#define WM8904_MIC_SHRT_EINT_POL_SHIFT 1 /* MIC_SHRT_EINT_POL */ +#define WM8904_MIC_SHRT_EINT_POL_WIDTH 1 /* MIC_SHRT_EINT_POL */ +#define WM8904_MIC_DET_EINT_POL 0x0001 /* MIC_DET_EINT_POL */ +#define WM8904_MIC_DET_EINT_POL_MASK 0x0001 /* MIC_DET_EINT_POL */ +#define WM8904_MIC_DET_EINT_POL_SHIFT 0 /* MIC_DET_EINT_POL */ +#define WM8904_MIC_DET_EINT_POL_WIDTH 1 /* MIC_DET_EINT_POL */ + +/* + * R130 (0x82) - Interrupt Debounce + */ +#define WM8904_GPIO_BCLK_EINT_DB 0x0200 /* GPIO_BCLK_EINT_DB */ +#define WM8904_GPIO_BCLK_EINT_DB_MASK 0x0200 /* GPIO_BCLK_EINT_DB */ +#define WM8904_GPIO_BCLK_EINT_DB_SHIFT 9 /* GPIO_BCLK_EINT_DB */ +#define WM8904_GPIO_BCLK_EINT_DB_WIDTH 1 /* GPIO_BCLK_EINT_DB */ +#define WM8904_WSEQ_EINT_DB 0x0100 /* WSEQ_EINT_DB */ +#define WM8904_WSEQ_EINT_DB_MASK 0x0100 /* WSEQ_EINT_DB */ +#define WM8904_WSEQ_EINT_DB_SHIFT 8 /* WSEQ_EINT_DB */ +#define WM8904_WSEQ_EINT_DB_WIDTH 1 /* WSEQ_EINT_DB */ +#define WM8904_GPIO3_EINT_DB 0x0080 /* GPIO3_EINT_DB */ +#define WM8904_GPIO3_EINT_DB_MASK 0x0080 /* GPIO3_EINT_DB */ +#define WM8904_GPIO3_EINT_DB_SHIFT 7 /* GPIO3_EINT_DB */ +#define WM8904_GPIO3_EINT_DB_WIDTH 1 /* GPIO3_EINT_DB */ +#define WM8904_GPIO2_EINT_DB 0x0040 /* GPIO2_EINT_DB */ +#define WM8904_GPIO2_EINT_DB_MASK 0x0040 /* GPIO2_EINT_DB */ +#define WM8904_GPIO2_EINT_DB_SHIFT 6 /* GPIO2_EINT_DB */ +#define WM8904_GPIO2_EINT_DB_WIDTH 1 /* GPIO2_EINT_DB */ +#define WM8904_GPIO1_EINT_DB 0x0020 /* GPIO1_EINT_DB */ +#define WM8904_GPIO1_EINT_DB_MASK 0x0020 /* GPIO1_EINT_DB */ +#define WM8904_GPIO1_EINT_DB_SHIFT 5 /* GPIO1_EINT_DB */ +#define WM8904_GPIO1_EINT_DB_WIDTH 1 /* GPIO1_EINT_DB */ +#define WM8904_GPI8_EINT_DB 0x0010 /* GPI8_EINT_DB */ +#define WM8904_GPI8_EINT_DB_MASK 0x0010 /* GPI8_EINT_DB */ +#define WM8904_GPI8_EINT_DB_SHIFT 4 /* GPI8_EINT_DB */ +#define WM8904_GPI8_EINT_DB_WIDTH 1 /* GPI8_EINT_DB */ +#define WM8904_GPI7_EINT_DB 0x0008 /* GPI7_EINT_DB */ +#define WM8904_GPI7_EINT_DB_MASK 0x0008 /* GPI7_EINT_DB */ +#define WM8904_GPI7_EINT_DB_SHIFT 3 /* GPI7_EINT_DB */ +#define WM8904_GPI7_EINT_DB_WIDTH 1 /* GPI7_EINT_DB */ +#define WM8904_FLL_LOCK_EINT_DB 0x0004 /* FLL_LOCK_EINT_DB */ +#define WM8904_FLL_LOCK_EINT_DB_MASK 0x0004 /* FLL_LOCK_EINT_DB */ +#define WM8904_FLL_LOCK_EINT_DB_SHIFT 2 /* FLL_LOCK_EINT_DB */ +#define WM8904_FLL_LOCK_EINT_DB_WIDTH 1 /* FLL_LOCK_EINT_DB */ +#define WM8904_MIC_SHRT_EINT_DB 0x0002 /* MIC_SHRT_EINT_DB */ +#define WM8904_MIC_SHRT_EINT_DB_MASK 0x0002 /* MIC_SHRT_EINT_DB */ +#define WM8904_MIC_SHRT_EINT_DB_SHIFT 1 /* MIC_SHRT_EINT_DB */ +#define WM8904_MIC_SHRT_EINT_DB_WIDTH 1 /* MIC_SHRT_EINT_DB */ +#define WM8904_MIC_DET_EINT_DB 0x0001 /* MIC_DET_EINT_DB */ +#define WM8904_MIC_DET_EINT_DB_MASK 0x0001 /* MIC_DET_EINT_DB */ +#define WM8904_MIC_DET_EINT_DB_SHIFT 0 /* MIC_DET_EINT_DB */ +#define WM8904_MIC_DET_EINT_DB_WIDTH 1 /* MIC_DET_EINT_DB */ + +/* + * R134 (0x86) - EQ1 + */ +#define WM8904_EQ_ENA 0x0001 /* EQ_ENA */ +#define WM8904_EQ_ENA_MASK 0x0001 /* EQ_ENA */ +#define WM8904_EQ_ENA_SHIFT 0 /* EQ_ENA */ +#define WM8904_EQ_ENA_WIDTH 1 /* EQ_ENA */ + +/* + * R135 (0x87) - EQ2 + */ +#define WM8904_EQ_B1_GAIN_MASK 0x001F /* EQ_B1_GAIN - [4:0] */ +#define WM8904_EQ_B1_GAIN_SHIFT 0 /* EQ_B1_GAIN - [4:0] */ +#define WM8904_EQ_B1_GAIN_WIDTH 5 /* EQ_B1_GAIN - [4:0] */ + +/* + * R136 (0x88) - EQ3 + */ +#define WM8904_EQ_B2_GAIN_MASK 0x001F /* EQ_B2_GAIN - [4:0] */ +#define WM8904_EQ_B2_GAIN_SHIFT 0 /* EQ_B2_GAIN - [4:0] */ +#define WM8904_EQ_B2_GAIN_WIDTH 5 /* EQ_B2_GAIN - [4:0] */ + +/* + * R137 (0x89) - EQ4 + */ +#define WM8904_EQ_B3_GAIN_MASK 0x001F /* EQ_B3_GAIN - [4:0] */ +#define WM8904_EQ_B3_GAIN_SHIFT 0 /* EQ_B3_GAIN - [4:0] */ +#define WM8904_EQ_B3_GAIN_WIDTH 5 /* EQ_B3_GAIN - [4:0] */ + +/* + * R138 (0x8A) - EQ5 + */ +#define WM8904_EQ_B4_GAIN_MASK 0x001F /* EQ_B4_GAIN - [4:0] */ +#define WM8904_EQ_B4_GAIN_SHIFT 0 /* EQ_B4_GAIN - [4:0] */ +#define WM8904_EQ_B4_GAIN_WIDTH 5 /* EQ_B4_GAIN - [4:0] */ + +/* + * R139 (0x8B) - EQ6 + */ +#define WM8904_EQ_B5_GAIN_MASK 0x001F /* EQ_B5_GAIN - [4:0] */ +#define WM8904_EQ_B5_GAIN_SHIFT 0 /* EQ_B5_GAIN - [4:0] */ +#define WM8904_EQ_B5_GAIN_WIDTH 5 /* EQ_B5_GAIN - [4:0] */ + +/* + * R140 (0x8C) - EQ7 + */ +#define WM8904_EQ_B1_A_MASK 0xFFFF /* EQ_B1_A - [15:0] */ +#define WM8904_EQ_B1_A_SHIFT 0 /* EQ_B1_A - [15:0] */ +#define WM8904_EQ_B1_A_WIDTH 16 /* EQ_B1_A - [15:0] */ + +/* + * R141 (0x8D) - EQ8 + */ +#define WM8904_EQ_B1_B_MASK 0xFFFF /* EQ_B1_B - [15:0] */ +#define WM8904_EQ_B1_B_SHIFT 0 /* EQ_B1_B - [15:0] */ +#define WM8904_EQ_B1_B_WIDTH 16 /* EQ_B1_B - [15:0] */ + +/* + * R142 (0x8E) - EQ9 + */ +#define WM8904_EQ_B1_PG_MASK 0xFFFF /* EQ_B1_PG - [15:0] */ +#define WM8904_EQ_B1_PG_SHIFT 0 /* EQ_B1_PG - [15:0] */ +#define WM8904_EQ_B1_PG_WIDTH 16 /* EQ_B1_PG - [15:0] */ + +/* + * R143 (0x8F) - EQ10 + */ +#define WM8904_EQ_B2_A_MASK 0xFFFF /* EQ_B2_A - [15:0] */ +#define WM8904_EQ_B2_A_SHIFT 0 /* EQ_B2_A - [15:0] */ +#define WM8904_EQ_B2_A_WIDTH 16 /* EQ_B2_A - [15:0] */ + +/* + * R144 (0x90) - EQ11 + */ +#define WM8904_EQ_B2_B_MASK 0xFFFF /* EQ_B2_B - [15:0] */ +#define WM8904_EQ_B2_B_SHIFT 0 /* EQ_B2_B - [15:0] */ +#define WM8904_EQ_B2_B_WIDTH 16 /* EQ_B2_B - [15:0] */ + +/* + * R145 (0x91) - EQ12 + */ +#define WM8904_EQ_B2_C_MASK 0xFFFF /* EQ_B2_C - [15:0] */ +#define WM8904_EQ_B2_C_SHIFT 0 /* EQ_B2_C - [15:0] */ +#define WM8904_EQ_B2_C_WIDTH 16 /* EQ_B2_C - [15:0] */ + +/* + * R146 (0x92) - EQ13 + */ +#define WM8904_EQ_B2_PG_MASK 0xFFFF /* EQ_B2_PG - [15:0] */ +#define WM8904_EQ_B2_PG_SHIFT 0 /* EQ_B2_PG - [15:0] */ +#define WM8904_EQ_B2_PG_WIDTH 16 /* EQ_B2_PG - [15:0] */ + +/* + * R147 (0x93) - EQ14 + */ +#define WM8904_EQ_B3_A_MASK 0xFFFF /* EQ_B3_A - [15:0] */ +#define WM8904_EQ_B3_A_SHIFT 0 /* EQ_B3_A - [15:0] */ +#define WM8904_EQ_B3_A_WIDTH 16 /* EQ_B3_A - [15:0] */ + +/* + * R148 (0x94) - EQ15 + */ +#define WM8904_EQ_B3_B_MASK 0xFFFF /* EQ_B3_B - [15:0] */ +#define WM8904_EQ_B3_B_SHIFT 0 /* EQ_B3_B - [15:0] */ +#define WM8904_EQ_B3_B_WIDTH 16 /* EQ_B3_B - [15:0] */ + +/* + * R149 (0x95) - EQ16 + */ +#define WM8904_EQ_B3_C_MASK 0xFFFF /* EQ_B3_C - [15:0] */ +#define WM8904_EQ_B3_C_SHIFT 0 /* EQ_B3_C - [15:0] */ +#define WM8904_EQ_B3_C_WIDTH 16 /* EQ_B3_C - [15:0] */ + +/* + * R150 (0x96) - EQ17 + */ +#define WM8904_EQ_B3_PG_MASK 0xFFFF /* EQ_B3_PG - [15:0] */ +#define WM8904_EQ_B3_PG_SHIFT 0 /* EQ_B3_PG - [15:0] */ +#define WM8904_EQ_B3_PG_WIDTH 16 /* EQ_B3_PG - [15:0] */ + +/* + * R151 (0x97) - EQ18 + */ +#define WM8904_EQ_B4_A_MASK 0xFFFF /* EQ_B4_A - [15:0] */ +#define WM8904_EQ_B4_A_SHIFT 0 /* EQ_B4_A - [15:0] */ +#define WM8904_EQ_B4_A_WIDTH 16 /* EQ_B4_A - [15:0] */ + +/* + * R152 (0x98) - EQ19 + */ +#define WM8904_EQ_B4_B_MASK 0xFFFF /* EQ_B4_B - [15:0] */ +#define WM8904_EQ_B4_B_SHIFT 0 /* EQ_B4_B - [15:0] */ +#define WM8904_EQ_B4_B_WIDTH 16 /* EQ_B4_B - [15:0] */ + +/* + * R153 (0x99) - EQ20 + */ +#define WM8904_EQ_B4_C_MASK 0xFFFF /* EQ_B4_C - [15:0] */ +#define WM8904_EQ_B4_C_SHIFT 0 /* EQ_B4_C - [15:0] */ +#define WM8904_EQ_B4_C_WIDTH 16 /* EQ_B4_C - [15:0] */ + +/* + * R154 (0x9A) - EQ21 + */ +#define WM8904_EQ_B4_PG_MASK 0xFFFF /* EQ_B4_PG - [15:0] */ +#define WM8904_EQ_B4_PG_SHIFT 0 /* EQ_B4_PG - [15:0] */ +#define WM8904_EQ_B4_PG_WIDTH 16 /* EQ_B4_PG - [15:0] */ + +/* + * R155 (0x9B) - EQ22 + */ +#define WM8904_EQ_B5_A_MASK 0xFFFF /* EQ_B5_A - [15:0] */ +#define WM8904_EQ_B5_A_SHIFT 0 /* EQ_B5_A - [15:0] */ +#define WM8904_EQ_B5_A_WIDTH 16 /* EQ_B5_A - [15:0] */ + +/* + * R156 (0x9C) - EQ23 + */ +#define WM8904_EQ_B5_B_MASK 0xFFFF /* EQ_B5_B - [15:0] */ +#define WM8904_EQ_B5_B_SHIFT 0 /* EQ_B5_B - [15:0] */ +#define WM8904_EQ_B5_B_WIDTH 16 /* EQ_B5_B - [15:0] */ + +/* + * R157 (0x9D) - EQ24 + */ +#define WM8904_EQ_B5_PG_MASK 0xFFFF /* EQ_B5_PG - [15:0] */ +#define WM8904_EQ_B5_PG_SHIFT 0 /* EQ_B5_PG - [15:0] */ +#define WM8904_EQ_B5_PG_WIDTH 16 /* EQ_B5_PG - [15:0] */ + +/* + * R161 (0xA1) - Control Interface Test 1 + */ +#define WM8904_USER_KEY 0x0002 /* USER_KEY */ +#define WM8904_USER_KEY_MASK 0x0002 /* USER_KEY */ +#define WM8904_USER_KEY_SHIFT 1 /* USER_KEY */ +#define WM8904_USER_KEY_WIDTH 1 /* USER_KEY */ + +/* + * R204 (0xCC) - Analogue Output Bias 0 + */ +#define WM8904_PGA_BIAS_MASK 0x0070 /* PGA_BIAS - [6:4] */ +#define WM8904_PGA_BIAS_SHIFT 4 /* PGA_BIAS - [6:4] */ +#define WM8904_PGA_BIAS_WIDTH 3 /* PGA_BIAS - [6:4] */ + +/* + * R247 (0xF7) - FLL NCO Test 0 + */ +#define WM8904_FLL_FRC_NCO 0x0001 /* FLL_FRC_NCO */ +#define WM8904_FLL_FRC_NCO_MASK 0x0001 /* FLL_FRC_NCO */ +#define WM8904_FLL_FRC_NCO_SHIFT 0 /* FLL_FRC_NCO */ +#define WM8904_FLL_FRC_NCO_WIDTH 1 /* FLL_FRC_NCO */ + +/* + * R248 (0xF8) - FLL NCO Test 1 + */ +#define WM8904_FLL_FRC_NCO_VAL_MASK 0x003F /* FLL_FRC_NCO_VAL - [5:0] */ +#define WM8904_FLL_FRC_NCO_VAL_SHIFT 0 /* FLL_FRC_NCO_VAL - [5:0] */ +#define WM8904_FLL_FRC_NCO_VAL_WIDTH 6 /* FLL_FRC_NCO_VAL - [5:0] */ + +#endif -- cgit v0.10.2 From ffbfd336f9eac361e1630cfcb17a70607551daf2 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 30 Nov 2009 17:56:11 +0100 Subject: ASoC: Add regulator support to CS4270 codec driver Signed-off-by: Daniel Mack Acked-by: Timur Tabi Cc: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ffe122d..8b54575 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -28,6 +28,7 @@ #include #include #include +#include #include "cs4270.h" @@ -106,6 +107,10 @@ #define CS4270_MUTE_DAC_A 0x01 #define CS4270_MUTE_DAC_B 0x02 +static const char *supply_names[] = { + "va", "vd", "vlc" +}; + /* Private data for the CS4270 */ struct cs4270_private { struct snd_soc_codec codec; @@ -114,6 +119,9 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; unsigned int manual_mute; + + /* power domain regulators */ + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; /** @@ -579,7 +587,8 @@ static int cs4270_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = cs4270_codec; - int ret; + struct cs4270_private *cs4270 = codec->private_data; + int i, ret; /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ socdev->card->codec = codec; @@ -599,6 +608,15 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } + /* get the power supply regulators */ + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + cs4270->supplies[i].supply = supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret < 0) + goto error_free_pcms; + return 0; error_free_pcms: @@ -616,8 +634,11 @@ error_free_pcms: static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return 0; }; @@ -799,17 +820,33 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { struct snd_soc_codec *codec = cs4270_codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + struct cs4270_private *cs4270 = codec->private_data; + int reg, ret; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + if (reg < 0) + return reg; + + ret = snd_soc_write(codec, CS4270_PWRCTL, reg); + if (ret < 0) + return ret; + + regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + + return 0; } static int cs4270_soc_resume(struct platform_device *pdev) { struct snd_soc_codec *codec = cs4270_codec; + struct cs4270_private *cs4270 = codec->private_data; struct i2c_client *i2c_client = codec->control_data; int reg; + regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ ndelay(500); -- cgit v0.10.2 From 14ff3e78304e3f7fe18f950c3aa0686e6800b3fb Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 10 Dec 2009 20:39:28 +0100 Subject: ALSA: dt019x: merge into the als100 driver The als100 driver is so similar to the dt019x/als007 driver that one driver's source can be used for both drivers with only few changes. Merge the dt019x driver into the als100. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 02fe81c..194af3b0 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -63,15 +63,16 @@ config SND_AD1848 will be called snd-ad1848. config SND_ALS100 - tristate "Avance Logic ALS100/ALS120" + tristate "Diamond Tech. DT-019x and Avance Logic ALSxxx" depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP help - Say Y here to include support for soundcards based on Avance - Logic ALS100, ALS110, ALS120 and ALS200 chips. + Say Y here to include support for soundcards based on the + Diamond Technologies DT-019X or Avance Logic chips: ALS007, + ALS100, ALS110, ALS120 and ALS200 chips. To compile this driver as a module, choose M here: the module will be called snd-als100. @@ -127,20 +128,6 @@ config SND_CS4236 To compile this driver as a module, choose M here: the module will be called snd-cs4236. -config SND_DT019X - tristate "Diamond Technologies DT-019X, Avance Logic ALS-007" - depends on PNP - select ISAPNP - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_SB16_DSP - help - Say Y here to include support for soundcards based on the - Diamond Technologies DT-019X or Avance Logic ALS-007 chips. - - To compile this driver as a module, choose M here: the module - will be called snd-dt019x. - config SND_ES968 tristate "Generic ESS ES968 driver" depends on PNP diff --git a/sound/isa/Makefile b/sound/isa/Makefile index b906b9a..c73d30c 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -7,7 +7,6 @@ snd-adlib-objs := adlib.o snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o snd-cmi8330-objs := cmi8330.o -snd-dt019x-objs := dt019x.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o @@ -19,7 +18,6 @@ obj-$(CONFIG_SND_ADLIB) += snd-adlib.o obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o -obj-$(CONFIG_SND_DT019X) += snd-dt019x.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 5fd52e4..20becc8 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -2,9 +2,13 @@ /* card-als100.c - driver for Avance Logic ALS100 based soundcards. Copyright (C) 1999-2000 by Massimo Piccioni + Copyright (C) 1999-2002 by Massimo Piccioni Thanks to Pierfrancesco 'qM2' Passerini. + Generalised for soundcards based on DT-0196 and ALS-007 chips + by Jonathan Woithe : June 2002. + This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or @@ -33,10 +37,10 @@ #define PFX "als100: " -MODULE_AUTHOR("Massimo Piccioni "); -MODULE_DESCRIPTION("Avance Logic ALS1X0"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," +MODULE_DESCRIPTION("Avance Logic ALS007/ALS1X0"); +MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," + "{Avance Logic ALS-007}}" + "{{Avance Logic,ALS100 - PRO16PNP}," "{Avance Logic,ALS110}," "{Avance Logic,ALS120}," "{Avance Logic,ALS200}," @@ -45,9 +49,12 @@ MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," "{Avance Logic,ALS120}," "{RTL,RTL3000}}"); +MODULE_AUTHOR("Massimo Piccioni "); +MODULE_LICENSE("GPL"); + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ @@ -57,14 +64,15 @@ static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for als100 based soundcard."); +MODULE_PARM_DESC(index, "Index value for Avance Logic based soundcard."); module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for als100 based soundcard."); +MODULE_PARM_DESC(id, "ID string for Avance Logic based soundcard."); module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable als100 based soundcard."); +MODULE_PARM_DESC(enable, "Enable Avance Logic based soundcard."); + +MODULE_ALIAS("snd-dt019x"); struct snd_card_als100 { - int dev_no; struct pnp_dev *dev; struct pnp_dev *devmpu; struct pnp_dev *devopl; @@ -72,25 +80,43 @@ struct snd_card_als100 { }; static struct pnp_card_device_id snd_als100_pnpids[] = { + /* DT197A30 */ + { .id = "RWB1688", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_DT019X }, + /* DT0196 / ALS-007 */ + { .id = "ALS0007", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_DT019X }, /* ALS100 - PRO16PNP */ - { .id = "ALS0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, + { .id = "ALS0001", + .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS110 - MF1000 - Digimate 3D Sound */ - { .id = "ALS0110", .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } } }, + { .id = "ALS0110", + .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS120 */ - { .id = "ALS0120", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } }, + { .id = "ALS0120", + .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS200 */ - { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } } }, + { .id = "ALS0200", + .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } }, + .driver_data = SB_HW_ALS100 }, /* ALS200 OEM */ - { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } } }, + { .id = "ALS0200", + .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } }, + .driver_data = SB_HW_ALS100 }, /* RTL3000 */ - { .id = "RTL3000", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } }, - { .id = "", } /* end */ + { .id = "RTL3000", + .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } }, + .driver_data = SB_HW_ALS100 }, + { .id = "" } /* end */ }; MODULE_DEVICE_TABLE(pnp_card, snd_als100_pnpids); -#define DRIVER_NAME "snd-card-als100" - static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, struct pnp_card_link *card, const struct pnp_card_device_id *id) @@ -113,8 +139,12 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, return err; } port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 1); - dma16[dev] = pnp_dma(pdev, 0); + if (id->driver_data == SB_HW_DT019X) + dma8[dev] = pnp_dma(pdev, 0); + else { + dma8[dev] = pnp_dma(pdev, 1); + dma16[dev] = pnp_dma(pdev, 0); + } irq[dev] = pnp_irq(pdev, 0); pdev = acard->devmpu; @@ -175,22 +205,33 @@ static int __devinit snd_card_als100_probe(int dev, } snd_card_set_dev(card, &pcard->card->dev); - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_sb16dsp_interrupt, - dma8[dev], - dma16[dev], - SB_HW_ALS100, &chip)) < 0) { + if (pid->driver_data == SB_HW_DT019X) + dma16[dev] = -1; + + error = snd_sbdsp_create(card, port[dev], irq[dev], + snd_sb16dsp_interrupt, + dma8[dev], dma16[dev], + pid->driver_data, + &chip); + if (error < 0) { snd_card_free(card); return error; } acard->chip = chip; - strcpy(card->driver, "ALS100"); - strcpy(card->shortname, "Avance Logic ALS100"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev], dma16[dev]); + if (pid->driver_data == SB_HW_DT019X) { + strcpy(card->driver, "DT-019X"); + strcpy(card->shortname, "Diamond Tech. DT-019X"); + sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", + card->shortname, chip->name, chip->port, + irq[dev], dma8[dev]); + } else { + strcpy(card->driver, "ALS100"); + strcpy(card->shortname, "Avance Logic ALS100"); + sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", + card->shortname, chip->name, chip->port, + irq[dev], dma8[dev], dma16[dev]); + } if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { snd_card_free(card); @@ -203,9 +244,19 @@ static int __devinit snd_card_als100_probe(int dev, } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if (snd_mpu401_uart_new(card, 0, MPU401_HW_ALS100, + int mpu_type = MPU401_HW_ALS100; + + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + + if (pid->driver_data == SB_HW_DT019X) + mpu_type = MPU401_HW_MPU401; + + if (snd_mpu401_uart_new(card, 0, + mpu_type, mpu_port[dev], 0, - mpu_irq[dev], IRQF_DISABLED, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } @@ -291,7 +342,7 @@ static int snd_als100_pnp_resume(struct pnp_card_link *pcard) static struct pnp_card_driver als100_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, - .name = "als100", + .name = "als100", .id_table = snd_als100_pnpids, .probe = snd_als100_pnp_detect, .remove = __devexit_p(snd_als100_pnp_remove), @@ -312,7 +363,7 @@ static int __init alsa_card_als100_init(void) if (!als100_devices) { pnp_unregister_card_driver(&als100_pnpc_driver); #ifdef MODULE - snd_printk(KERN_ERR "no ALS100 based soundcards found\n"); + snd_printk(KERN_ERR "no Avance Logic based soundcards found\n"); #endif return -ENODEV; } diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c deleted file mode 100644 index 80f5b1a..0000000 --- a/sound/isa/dt019x.c +++ /dev/null @@ -1,321 +0,0 @@ - -/* - dt019x.c - driver for Diamond Technologies DT-0197H based soundcards. - Copyright (C) 1999, 2002 by Massimo Piccioni - - Generalised for soundcards based on DT-0196 and ALS-007 chips - by Jonathan Woithe : June 2002. - - This program is free software; you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation; either version 2 of the License, or - (at your option) any later version. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with this program; if not, write to the Free Software - Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -*/ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#define PFX "dt019x: " - -MODULE_AUTHOR("Massimo Piccioni "); -MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," - "{Avance Logic ALS-007}}"); - -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ -static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */ -static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */ -static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ - -module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for DT-019X based soundcard."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for DT-019X based soundcard."); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable DT-019X based soundcard."); - -struct snd_card_dt019x { - struct pnp_dev *dev; - struct pnp_dev *devmpu; - struct pnp_dev *devopl; - struct snd_sb *chip; -}; - -static struct pnp_card_device_id snd_dt019x_pnpids[] = { - /* DT197A30 */ - { .id = "RWB1688", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, - /* DT0196 / ALS-007 */ - { .id = "ALS0007", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } }, - { .id = "", } -}; - -MODULE_DEVICE_TABLE(pnp_card, snd_dt019x_pnpids); - - -#define DRIVER_NAME "snd-card-dt019x" - - -static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard, - struct pnp_card_link *card, - const struct pnp_card_device_id *pid) -{ - struct pnp_dev *pdev; - int err; - - acard->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (acard->dev == NULL) - return -ENODEV; - - acard->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - acard->devopl = pnp_request_card_device(card, pid->devs[2].id, NULL); - - pdev = acard->dev; - - err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR PFX "DT-019X AUDIO pnp configure failure\n"); - return err; - } - - port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 0); - irq[dev] = pnp_irq(pdev, 0); - snd_printdd("dt019x: found audio interface: port=0x%lx, irq=0x%x, dma=0x%x\n", - port[dev],irq[dev],dma8[dev]); - - pdev = acard->devmpu; - if (pdev != NULL) { - err = pnp_activate_dev(pdev); - if (err < 0) { - pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "DT-019X MPU401 pnp configure failure, skipping\n"); - goto __mpu_error; - } - mpu_port[dev] = pnp_port_start(pdev, 0); - mpu_irq[dev] = pnp_irq(pdev, 0); - snd_printdd("dt019x: found MPU-401: port=0x%lx, irq=0x%x\n", - mpu_port[dev],mpu_irq[dev]); - } else { - __mpu_error: - acard->devmpu = NULL; - mpu_port[dev] = -1; - } - - pdev = acard->devopl; - if (pdev != NULL) { - err = pnp_activate_dev(pdev); - if (err < 0) { - pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "DT-019X OPL3 pnp configure failure, skipping\n"); - goto __fm_error; - } - fm_port[dev] = pnp_port_start(pdev, 0); - snd_printdd("dt019x: found OPL3 synth: port=0x%lx\n",fm_port[dev]); - } else { - __fm_error: - acard->devopl = NULL; - fm_port[dev] = -1; - } - - return 0; -} - -static int __devinit snd_card_dt019x_probe(int dev, struct pnp_card_link *pcard, const struct pnp_card_device_id *pid) -{ - int error; - struct snd_sb *chip; - struct snd_card *card; - struct snd_card_dt019x *acard; - struct snd_opl3 *opl3; - - error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_dt019x), &card); - if (error < 0) - return error; - acard = card->private_data; - - snd_card_set_dev(card, &pcard->card->dev); - if ((error = snd_card_dt019x_pnp(dev, acard, pcard, pid))) { - snd_card_free(card); - return error; - } - - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_sb16dsp_interrupt, - dma8[dev], - -1, - SB_HW_DT019X, - &chip)) < 0) { - snd_card_free(card); - return error; - } - acard->chip = chip; - - strcpy(card->driver, "DT-019X"); - strcpy(card->shortname, "Diamond Tech. DT-019X"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev]); - - if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_sbmixer_new(chip)) < 0) { - snd_card_free(card); - return error; - } - - if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if (mpu_irq[dev] == SNDRV_AUTO_IRQ) - mpu_irq[dev] = -1; - if (snd_mpu401_uart_new(card, 0, -/* MPU401_HW_SB,*/ - MPU401_HW_MPU401, - mpu_port[dev], 0, - mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, - NULL) < 0) - snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx ?\n", mpu_port[dev]); - } - - if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { - if (snd_opl3_create(card, - fm_port[dev], - fm_port[dev] + 2, - OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx ?\n", - fm_port[dev], fm_port[dev] + 2); - } else { - if ((error = snd_opl3_timer_new(opl3, 0, 1)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { - snd_card_free(card); - return error; - } - } - } - - if ((error = snd_card_register(card)) < 0) { - snd_card_free(card); - return error; - } - pnp_set_card_drvdata(pcard, card); - return 0; -} - -static unsigned int __devinitdata dt019x_devices; - -static int __devinit snd_dt019x_pnp_probe(struct pnp_card_link *card, - const struct pnp_card_device_id *pid) -{ - static int dev; - int res; - - for ( ; dev < SNDRV_CARDS; dev++) { - if (!enable[dev]) - continue; - res = snd_card_dt019x_probe(dev, card, pid); - if (res < 0) - return res; - dev++; - dt019x_devices++; - return 0; - } - return -ENODEV; -} - -static void __devexit snd_dt019x_pnp_remove(struct pnp_card_link * pcard) -{ - snd_card_free(pnp_get_card_drvdata(pcard)); - pnp_set_card_drvdata(pcard, NULL); -} - -#ifdef CONFIG_PM -static int snd_dt019x_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_dt019x *acard = card->private_data; - struct snd_sb *chip = acard->chip; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); - snd_sbmixer_suspend(chip); - return 0; -} - -static int snd_dt019x_pnp_resume(struct pnp_card_link *pcard) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_dt019x *acard = card->private_data; - struct snd_sb *chip = acard->chip; - - snd_sbdsp_reset(chip); - snd_sbmixer_resume(chip); - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif - -static struct pnp_card_driver dt019x_pnpc_driver = { - .flags = PNP_DRIVER_RES_DISABLE, - .name = "dt019x", - .id_table = snd_dt019x_pnpids, - .probe = snd_dt019x_pnp_probe, - .remove = __devexit_p(snd_dt019x_pnp_remove), -#ifdef CONFIG_PM - .suspend = snd_dt019x_pnp_suspend, - .resume = snd_dt019x_pnp_resume, -#endif -}; - -static int __init alsa_card_dt019x_init(void) -{ - int err; - - err = pnp_register_card_driver(&dt019x_pnpc_driver); - if (err) - return err; - - if (!dt019x_devices) { - pnp_unregister_card_driver(&dt019x_pnpc_driver); -#ifdef MODULE - snd_printk(KERN_ERR "no DT-019X / ALS-007 based soundcards found\n"); -#endif - return -ENODEV; - } - return 0; -} - -static void __exit alsa_card_dt019x_exit(void) -{ - pnp_unregister_card_driver(&dt019x_pnpc_driver); -} - -module_init(alsa_card_dt019x_init) -module_exit(alsa_card_dt019x_exit) -- cgit v0.10.2 From b2e8d7dab9d82be3851b8cbcc1ab64b1b2575844 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 10 Dec 2009 20:40:18 +0100 Subject: ALSA: opti93x: move controls definitions to opti93x driver Move OPTi93x controls definitions to the opti93x driver from the common wss-lib library module. These controls are used only by the opti93x driver. Also, fix capture source names. They are the same as opl3sa2 names. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 106be6e..ea4a671 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include #include @@ -546,6 +547,85 @@ __skip_mpu: #ifdef OPTi93X +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); + +static struct snd_kcontrol_new snd_opti93x_controls[] = { +WSS_DOUBLE("Master Playback Switch", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, + db_scale_6bit), +WSS_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +WSS_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +WSS_DOUBLE("Mic Playback Switch", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE("Aux Playback Switch", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +}; + +static int __devinit snd_opti93x_mixer(struct snd_wss *chip) +{ + struct snd_card *card; + unsigned int idx; + struct snd_ctl_elem_id id1, id2; + int err; + + if (snd_BUG_ON(!chip || !chip->pcm)) + return -EINVAL; + + card = chip->card; + + strcpy(card->mixername, chip->pcm->name); + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* reassign AUX0 switch to CD */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "Cannot rename opti93x control\n"); + return err; + } + /* reassign AUX1 switch to FM */ + strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; + strcpy(id2.name, "FM Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "Cannot rename opti93x control\n"); + return err; + } + /* remove AUX1 volume */ + strcpy(id1.name, "Aux Playback Volume"); id1.index = 1; + snd_ctl_remove_id(card, &id1); + + /* Replace WSS volume controls with OPTi93x volume controls */ + id1.index = 0; + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + strcpy(id1.name, snd_opti93x_controls[idx].name); + snd_ctl_remove_id(card, &id1); + + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], chip)); + if (err < 0) + return err; + } + return 0; +} + static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { struct snd_wss *codec = dev_id; @@ -752,6 +832,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) error = snd_wss_mixer(codec); if (error < 0) return error; +#ifdef OPTi93X + error = snd_opti93x_mixer(codec); + if (error < 0) + return error; +#endif #ifdef CS4231 error = snd_wss_timer(codec, 0, &timer); if (error < 0) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 5b9d6c1..9191b32 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2014,6 +2014,7 @@ static int snd_wss_info_mux(struct snd_kcontrol *kcontrol, case WSS_HW_INTERWAVE: ptexts = gusmax_texts; break; + case WSS_HW_OPTI93X: case WSS_HW_OPL3SA2: ptexts = opl3sa_texts; break; @@ -2246,54 +2247,12 @@ WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), }; -static struct snd_kcontrol_new snd_opti93x_controls[] = { -WSS_DOUBLE("Master Playback Switch", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, - db_scale_6bit), -WSS_DOUBLE("PCM Playback Switch", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -WSS_DOUBLE("FM Playback Switch", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Line Playback Switch", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), -WSS_DOUBLE("Mic Playback Switch", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Mic Playback Volume", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -WSS_DOUBLE("CD Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Aux Playback Switch", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_wss_info_mux, - .get = snd_wss_get_mux, - .put = snd_wss_put_mux, -} -}; - int snd_wss_mixer(struct snd_wss *chip) { struct snd_card *card; unsigned int idx; int err; + int count = ARRAY_SIZE(snd_wss_controls); if (snd_BUG_ON(!chip || !chip->pcm)) return -EINVAL; @@ -2302,28 +2261,19 @@ int snd_wss_mixer(struct snd_wss *chip) strcpy(card->mixername, chip->pcm->name); - if (chip->hardware == WSS_HW_OPTI93X) - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_opti93x_controls[idx], - chip)); - if (err < 0) - return err; - } - else { - int count = ARRAY_SIZE(snd_wss_controls); - - /* Use only the first 11 entries on AD1848 */ - if (chip->hardware & WSS_HW_AD1848_MASK) - count = 11; - - for (idx = 0; idx < count; idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_wss_controls[idx], - chip)); - if (err < 0) - return err; - } + /* Use only the first 11 entries on AD1848 */ + if (chip->hardware & WSS_HW_AD1848_MASK) + count = 11; + /* There is no loopback on OPTI93X */ + else if (chip->hardware == WSS_HW_OPTI93X) + count = 9; + + for (idx = 0; idx < count; idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_wss_controls[idx], + chip)); + if (err < 0) + return err; } return 0; } -- cgit v0.10.2 From e9d0a803c127e2e30afb0df780ccb3af4e2adb28 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 12 Dec 2009 09:51:03 +0100 Subject: ALSA: opti93x: use dB scale for mixer controls Add dB scale for mixer controls. Fix dB scale for Master Volume control. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index ea4a671..b0ea310 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -547,32 +547,40 @@ __skip_mpu: #ifdef OPTi93X -static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_3db_step, -9300, 300, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit_12db_max, -3300, 300, 0); static struct snd_kcontrol_new snd_opti93x_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), WSS_DOUBLE_TLV("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, - db_scale_6bit), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -WSS_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), + db_scale_5bit_3db_step), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1, + db_scale_5bit), +WSS_DOUBLE_TLV("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Mic Playback Volume", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE_TLV("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), +WSS_DOUBLE_TLV("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1, + db_scale_4bit_12db_max), }; static int __devinit snd_opti93x_mixer(struct snd_wss *chip) -- cgit v0.10.2 From 74c2b45b714e49b427584b4bd8f44f1a24d82d9c Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 13 Dec 2009 21:13:44 +0100 Subject: ALSA: sb_mixer: convert pointer tables to mixer control tables Convert table of pointers to mixer controls into tables of the mixer controls. It saves about 20% of the snd-sb-common module size reported by lsmod. The als4000 uses part of sb16's control table. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 318ff0c..8cfc41f 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -528,20 +528,11 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty * SB 2.0 specific mixer elements */ -static struct sbmix_elem snd_sb20_ctl_master_play_vol = - SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7); -static struct sbmix_elem snd_sb20_ctl_pcm_play_vol = - SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3); -static struct sbmix_elem snd_sb20_ctl_synth_play_vol = - SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7); -static struct sbmix_elem snd_sb20_ctl_cd_play_vol = - SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7); - -static struct sbmix_elem *snd_sb20_controls[] = { - &snd_sb20_ctl_master_play_vol, - &snd_sb20_ctl_pcm_play_vol, - &snd_sb20_ctl_synth_play_vol, - &snd_sb20_ctl_cd_play_vol +static struct sbmix_elem snd_sb20_controls[] = { + SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7), + SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3), + SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7), + SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7) }; static unsigned char snd_sb20_init_values[][2] = { @@ -552,41 +543,24 @@ static unsigned char snd_sb20_init_values[][2] = { /* * SB Pro specific mixer elements */ -static struct sbmix_elem snd_sbpro_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_pcm_play_filter = - SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1); -static struct sbmix_elem snd_sbpro_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3); -static struct sbmix_elem snd_sbpro_ctl_capture_source = +static struct sbmix_elem snd_sbpro_controls[] = { + SB_DOUBLE("Master Playback Volume", + SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7), + SB_DOUBLE("PCM Playback Volume", + SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7), + SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1), + SB_DOUBLE("Synth Playback Volume", + SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7), + SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7), + SB_DOUBLE("Line Playback Volume", + SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7), + SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3), { .name = "Capture Source", .type = SB_MIX_CAPTURE_PRO - }; -static struct sbmix_elem snd_sbpro_ctl_capture_filter = - SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1); -static struct sbmix_elem snd_sbpro_ctl_capture_low_filter = - SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1); - -static struct sbmix_elem *snd_sbpro_controls[] = { - &snd_sbpro_ctl_master_play_vol, - &snd_sbpro_ctl_pcm_play_vol, - &snd_sbpro_ctl_pcm_play_filter, - &snd_sbpro_ctl_synth_play_vol, - &snd_sbpro_ctl_cd_play_vol, - &snd_sbpro_ctl_line_play_vol, - &snd_sbpro_ctl_mic_play_vol, - &snd_sbpro_ctl_capture_source, - &snd_sbpro_ctl_capture_filter, - &snd_sbpro_ctl_capture_low_filter + }, + SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1), + SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1) }; static unsigned char snd_sbpro_init_values[][2] = { @@ -598,68 +572,42 @@ static unsigned char snd_sbpro_init_values[][2] = { /* * SB16 specific mixer elements */ -static struct sbmix_elem snd_sb16_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_3d_enhance_switch = - SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1); -static struct sbmix_elem snd_sb16_ctl_tone_bass = - SB_DOUBLE("Tone Control - Bass", SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15); -static struct sbmix_elem snd_sb16_ctl_tone_treble = - SB_DOUBLE("Tone Control - Treble", SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15); -static struct sbmix_elem snd_sb16_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_synth_capture_route = - SB16_INPUT_SW("Synth Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5); -static struct sbmix_elem snd_sb16_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_cd_capture_route = - SB16_INPUT_SW("CD Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1); -static struct sbmix_elem snd_sb16_ctl_cd_play_switch = - SB_DOUBLE("CD Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1); -static struct sbmix_elem snd_sb16_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_line_capture_route = - SB16_INPUT_SW("Line Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3); -static struct sbmix_elem snd_sb16_ctl_line_play_switch = - SB_DOUBLE("Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1); -static struct sbmix_elem snd_sb16_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_mic_capture_route = - SB16_INPUT_SW("Mic Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0); -static struct sbmix_elem snd_sb16_ctl_mic_play_switch = - SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1); -static struct sbmix_elem snd_sb16_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31); -static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol = - SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3); -static struct sbmix_elem snd_sb16_ctl_capture_vol = - SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3); -static struct sbmix_elem snd_sb16_ctl_play_vol = - SB_DOUBLE("Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3); -static struct sbmix_elem snd_sb16_ctl_auto_mic_gain = - SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1); - -static struct sbmix_elem *snd_sb16_controls[] = { - &snd_sb16_ctl_master_play_vol, - &snd_sb16_ctl_3d_enhance_switch, - &snd_sb16_ctl_tone_bass, - &snd_sb16_ctl_tone_treble, - &snd_sb16_ctl_pcm_play_vol, - &snd_sb16_ctl_synth_capture_route, - &snd_sb16_ctl_synth_play_vol, - &snd_sb16_ctl_cd_capture_route, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_cd_play_vol, - &snd_sb16_ctl_line_capture_route, - &snd_sb16_ctl_line_play_switch, - &snd_sb16_ctl_line_play_vol, - &snd_sb16_ctl_mic_capture_route, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_mic_play_vol, - &snd_sb16_ctl_pc_speaker_vol, - &snd_sb16_ctl_capture_vol, - &snd_sb16_ctl_play_vol, - &snd_sb16_ctl_auto_mic_gain +static struct sbmix_elem snd_sb16_controls[] = { + SB_DOUBLE("Master Playback Volume", + SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31), + SB_DOUBLE("PCM Playback Volume", + SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("Synth Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5), + SB_DOUBLE("Synth Playback Volume", + SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("CD Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1), + SB_DOUBLE("CD Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1), + SB_DOUBLE("CD Playback Volume", + SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("Mic Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0), + SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), + SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), + SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3), + SB_DOUBLE("Capture Volume", + SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3), + SB_DOUBLE("Playback Volume", + SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3), + SB16_INPUT_SW("Line Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3), + SB_DOUBLE("Line Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1), + SB_DOUBLE("Line Playback Volume", + SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31), + SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1), + SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1), + SB_DOUBLE("Tone Control - Bass", + SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15), + SB_DOUBLE("Tone Control - Treble", + SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15) }; static unsigned char snd_sb16_init_values[][2] = { @@ -678,46 +626,34 @@ static unsigned char snd_sb16_init_values[][2] = { /* * DT019x specific mixer elements */ -static struct sbmix_elem snd_dt019x_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7); -static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol = - SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7); -static struct sbmix_elem snd_dt019x_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch = - SB_DOUBLE("PCM Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2,1, 1); -static struct sbmix_elem snd_dt019x_ctl_synth_play_switch = - SB_DOUBLE("Synth Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4,3, 1); -static struct sbmix_elem snd_dt019x_ctl_capture_source = +static struct sbmix_elem snd_dt019x_controls[] = { + /* ALS4000 below has some parts which we might be lacking, + * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ + SB_DOUBLE("Master Playback Volume", + SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4, 0, 15), + SB_DOUBLE("PCM Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1), + SB_DOUBLE("PCM Playback Volume", + SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4, 0, 15), + SB_DOUBLE("Synth Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1), + SB_DOUBLE("Synth Playback Volume", + SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4, 0, 15), + SB_DOUBLE("CD Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1), + SB_DOUBLE("CD Playback Volume", + SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4, 0, 15), + SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), + SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7), + SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7), + SB_DOUBLE("Line Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1), + SB_DOUBLE("Line Playback Volume", + SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4, 0, 15), { .name = "Capture Source", .type = SB_MIX_CAPTURE_DT019X - }; - -static struct sbmix_elem *snd_dt019x_controls[] = { - /* ALS4000 below has some parts which we might be lacking, - * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ - &snd_dt019x_ctl_master_play_vol, - &snd_dt019x_ctl_pcm_play_vol, - &snd_dt019x_ctl_synth_play_vol, - &snd_dt019x_ctl_cd_play_vol, - &snd_dt019x_ctl_mic_play_vol, - &snd_dt019x_ctl_pc_speaker_vol, - &snd_dt019x_ctl_line_play_vol, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_line_play_switch, - &snd_dt019x_ctl_pcm_play_switch, - &snd_dt019x_ctl_synth_play_switch, - &snd_dt019x_ctl_capture_source + } }; static unsigned char snd_dt019x_init_values[][2] = { @@ -735,82 +671,37 @@ static unsigned char snd_dt019x_init_values[][2] = { /* * ALS4000 specific mixer elements */ -static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch = - SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1); -static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = { +static struct sbmix_elem snd_als4000_controls[] = { + SB_DOUBLE("PCM Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1), + SB_DOUBLE("Synth Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1), + SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03), + SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1), + { .name = "Master Mono Capture Route", .type = SB_MIX_MONO_CAPTURE_ALS4K - }; -static struct sbmix_elem snd_als4000_ctl_mono_playback_switch = - SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1); -static struct sbmix_elem snd_als4000_ctl_mic_20db_boost = - SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03); -static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback = - SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); -static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback = + }, + SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1), + SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01), + SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01), SB_SINGLE("Digital Loopback Switch", - SB_ALS4000_CR3_CONFIGURATION, 7, 0x01); -/* FIXME: functionality of 3D controls might be swapped, I didn't find - * a description of how to identify what is supposed to be what */ -static struct sbmix_elem snd_als4000_3d_control_switch = - SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01); -static struct sbmix_elem snd_als4000_3d_control_ratio = - SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07); -static struct sbmix_elem snd_als4000_3d_control_freq = + SB_ALS4000_CR3_CONFIGURATION, 7, 0x01), + /* FIXME: functionality of 3D controls might be swapped, I didn't find + * a description of how to identify what is supposed to be what */ + SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07), /* FIXME: maybe there's actually some standard 3D ctrl name for it?? */ - SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03); -static struct sbmix_elem snd_als4000_3d_control_delay = + SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03), /* FIXME: ALS4000a.pdf mentions BBD (Bucket Brigade Device) time delay, * but what ALSA 3D attribute is that actually? "Center", "Depth", * "Wide" or "Space" or even "Level"? Assuming "Wide" for now... */ - SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f); -static struct sbmix_elem snd_als4000_3d_control_poweroff_switch = - SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01); -static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch = + SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f), + SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01), SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch", - SB_ALS4000_FMDAC, 5, 0x01); + SB_ALS4000_FMDAC, 5, 0x01), #ifdef NOT_AVAILABLE -static struct sbmix_elem snd_als4000_ctl_fmdac = - SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01); -static struct sbmix_elem snd_als4000_ctl_qsound = - SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f); -#endif - -static struct sbmix_elem *snd_als4000_controls[] = { - /* ALS4000a.PDF regs page */ - &snd_sb16_ctl_master_play_vol, /* MX30/31 12 */ - &snd_dt019x_ctl_pcm_play_switch, /* MX4C 16 */ - &snd_sb16_ctl_pcm_play_vol, /* MX32/33 12 */ - &snd_sb16_ctl_synth_capture_route, /* MX3D/3E 14 */ - &snd_dt019x_ctl_synth_play_switch, /* MX4C 16 */ - &snd_sb16_ctl_synth_play_vol, /* MX34/35 12/13 */ - &snd_sb16_ctl_cd_capture_route, /* MX3D/3E 14 */ - &snd_sb16_ctl_cd_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_cd_play_vol, /* MX36/37 13 */ - &snd_sb16_ctl_line_capture_route, /* MX3D/3E 14 */ - &snd_sb16_ctl_line_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_line_play_vol, /* MX38/39 13 */ - &snd_sb16_ctl_mic_capture_route, /* MX3D/3E 14 */ - &snd_als4000_ctl_mic_20db_boost, /* MX4D 16 */ - &snd_sb16_ctl_mic_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_mic_play_vol, /* MX3A 13 */ - &snd_sb16_ctl_pc_speaker_vol, /* MX3B 14 */ - &snd_sb16_ctl_capture_vol, /* MX3F/40 15 */ - &snd_sb16_ctl_play_vol, /* MX41/42 15 */ - &snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */ - &snd_als4k_ctl_master_mono_capture_route, /* MX4B 16 */ - &snd_als4000_ctl_mono_playback_switch, /* MX4C 16 */ - &snd_als4000_ctl_mixer_analog_loopback, /* MX4D 16 */ - &snd_als4000_ctl_mixer_digital_loopback, /* CR3 21 */ - &snd_als4000_3d_control_switch, /* MX50 17 */ - &snd_als4000_3d_control_ratio, /* MX50 17 */ - &snd_als4000_3d_control_freq, /* MX50 17 */ - &snd_als4000_3d_control_delay, /* MX51 18 */ - &snd_als4000_3d_control_poweroff_switch, /* MX51 18 */ - &snd_als4000_ctl_3db_freq_control_switch, /* MX4F 17 */ -#ifdef NOT_AVAILABLE - &snd_als4000_ctl_fmdac, - &snd_als4000_ctl_qsound, + SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01), + SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f), #endif }; @@ -829,11 +720,10 @@ static unsigned char snd_als4000_init_values[][2] = { { SB_ALS4000_MIC_IN_GAIN, 0 }, }; - /* */ static int snd_sbmixer_init(struct snd_sb *chip, - struct sbmix_elem **controls, + struct sbmix_elem *controls, int controls_count, unsigned char map[][2], int map_count, @@ -856,7 +746,8 @@ static int snd_sbmixer_init(struct snd_sb *chip, } for (idx = 0; idx < controls_count; idx++) { - if ((err = snd_sbmixer_add_ctl_elem(chip, controls[idx])) < 0) + err = snd_sbmixer_add_ctl_elem(chip, &controls[idx]); + if (err < 0) return err; } snd_component_add(card, name); @@ -908,6 +799,15 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_ALS4000: + /* use only the first 16 controls from SB16 */ + err = snd_sbmixer_init(chip, + snd_sb16_controls, + 16, + snd_sb16_init_values, + ARRAY_SIZE(snd_sb16_init_values), + "ALS4000"); + if (err < 0) + return err; if ((err = snd_sbmixer_init(chip, snd_als4000_controls, ARRAY_SIZE(snd_als4000_controls), -- cgit v0.10.2 From 63978ab3e3e963db28093b53bb4598f2702e1ad7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 14 Dec 2009 12:48:35 +0100 Subject: sound: add Edirol UA-101 support Add experimental support for the Edirol UA-101 audio/MIDI interface. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8923597..7a0a4a9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1791,6 +1791,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. + Module snd-ua101 + ---------------- + + Module for the Edirol UA-101 audio/MIDI interface. + + This module supports multiple devices, autoprobe and hotplugging. + Module snd-usb-audio -------------------- diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 73525c0..8c29258 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -21,6 +21,18 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_UA101 + tristate "Edirol UA-101 driver (EXPERIMENTAL)" + depends on EXPERIMENTAL + select SND_PCM + select SND_RAWMIDI + help + Say Y here to include support for the Edirol UA-101 audio/MIDI + interface. + + To compile this driver as a module, choose M here: the module + will be called snd-ua101. + config SND_USB_USX2Y tristate "Tascam US-122, US-224 and US-428 USB driver" depends on X86 || PPC || ALPHA diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abb288b..5bf64ae 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -4,9 +4,11 @@ snd-usb-audio-objs := usbaudio.o usbmixer.o snd-usb-lib-objs := usbmidi.o +snd-ua101-objs := ua101.o # Toplevel Module Dependency obj-$(CONFIG_SND_USB_AUDIO) += snd-usb-audio.o snd-usb-lib.o +obj-$(CONFIG_SND_USB_UA101) += snd-ua101.o snd-usb-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usb-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usb-lib.o diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c new file mode 100644 index 0000000..ab9f8a2 --- /dev/null +++ b/sound/usb/ua101.c @@ -0,0 +1,1457 @@ +/* + * Edirol UA-101 driver + * Copyright (c) Clemens Ladisch + * + * This driver is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver. If not, see . + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "usbaudio.h" + +MODULE_DESCRIPTION("Edirol UA-101 driver"); +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101}}"); + +/* I use my UA-1A for testing because I don't have a UA-101 ... */ +#define UA1A_HACK + +/* + * Should not be lower than the minimum scheduling delay of the host + * controller. Some Intel controllers need more than one frame; as long as + * that driver doesn't tell us about this, use 1.5 frames just to be sure. + */ +#define MIN_QUEUE_LENGTH 12 +/* Somewhat random. */ +#define MAX_QUEUE_LENGTH 30 +/* + * This magic value optimizes memory usage efficiency for the UA-101's packet + * sizes at all sample rates, taking into account the stupid cache pool sizes + * that usb_buffer_alloc() uses. + */ +#define DEFAULT_QUEUE_LENGTH 21 + +#define MAX_PACKET_SIZE 672 /* hardware specific */ +#define MAX_MEMORY_BUFFERS DIV_ROUND_UP(MAX_QUEUE_LENGTH, \ + PAGE_SIZE / MAX_PACKET_SIZE) + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static unsigned int queue_length = 21; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "card index"); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string"); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "enable card"); +module_param(queue_length, uint, 0644); +MODULE_PARM_DESC(queue_length, "USB queue length in microframes, " + __stringify(MIN_QUEUE_LENGTH)"-"__stringify(MAX_QUEUE_LENGTH)); + +enum { + INTF_PLAYBACK, + INTF_CAPTURE, + INTF_MIDI, + + INTF_COUNT +}; + +/* bits in struct ua101::states */ +enum { + USB_CAPTURE_RUNNING, + USB_PLAYBACK_RUNNING, + ALSA_CAPTURE_OPEN, + ALSA_PLAYBACK_OPEN, + ALSA_CAPTURE_RUNNING, + ALSA_PLAYBACK_RUNNING, + CAPTURE_URB_COMPLETED, + PLAYBACK_URB_COMPLETED, + DISCONNECTED, +}; + +struct ua101 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf[INTF_COUNT]; + int card_index; + struct snd_pcm *pcm; + struct list_head midi_list; + u64 format_bit; + unsigned int rate; + unsigned int packets_per_second; + spinlock_t lock; + struct mutex mutex; + unsigned long states; + + /* FIFO to synchronize playback rate to capture rate */ + unsigned int rate_feedback_start; + unsigned int rate_feedback_count; + u8 rate_feedback[MAX_QUEUE_LENGTH]; + + struct list_head ready_playback_urbs; + struct tasklet_struct playback_tasklet; + wait_queue_head_t alsa_capture_wait; + wait_queue_head_t rate_feedback_wait; + wait_queue_head_t alsa_playback_wait; + struct ua101_stream { + struct snd_pcm_substream *substream; + unsigned int usb_pipe; + unsigned int channels; + unsigned int frame_bytes; + unsigned int max_packet_bytes; + unsigned int period_pos; + unsigned int buffer_pos; + unsigned int queue_length; + struct ua101_urb { + struct urb urb; + struct usb_iso_packet_descriptor iso_frame_desc[1]; + struct list_head ready_list; + } *urbs[MAX_QUEUE_LENGTH]; + struct { + unsigned int size; + void *addr; + dma_addr_t dma; + } buffers[MAX_MEMORY_BUFFERS]; + } capture, playback; + + unsigned int fps[10]; + unsigned int frame_counter; +}; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver ua101_driver; + +static void abort_alsa_playback(struct ua101 *ua); +static void abort_alsa_capture(struct ua101 *ua); + +/* allocate virtual buffer; may be called more than once */ +static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, + size_t size) +{ + struct snd_pcm_runtime *runtime = subs->runtime; + + if (runtime->dma_area) { + if (runtime->dma_bytes >= size) + return 0; /* already large enough */ + vfree(runtime->dma_area); + } + runtime->dma_area = vmalloc_user(size); + if (!runtime->dma_area) + return -ENOMEM; + runtime->dma_bytes = size; + return 0; +} + +/* free virtual buffer; may be called more than once */ +static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) +{ + struct snd_pcm_runtime *runtime = subs->runtime; + + vfree(runtime->dma_area); + runtime->dma_area = NULL; + return 0; +} + +/* get the physical page pointer at the given offset */ +static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, + unsigned long offset) +{ + void *pageptr = subs->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); +} + +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} + +static void abort_usb_capture(struct ua101 *ua) +{ + if (test_and_clear_bit(USB_CAPTURE_RUNNING, &ua->states)) { + wake_up(&ua->alsa_capture_wait); + wake_up(&ua->rate_feedback_wait); + } +} + +static void abort_usb_playback(struct ua101 *ua) +{ + if (test_and_clear_bit(USB_PLAYBACK_RUNNING, &ua->states)) + wake_up(&ua->alsa_playback_wait); +} + +static void playback_urb_complete(struct urb *usb_urb) +{ + struct ua101_urb *urb = (struct ua101_urb *)usb_urb; + struct ua101 *ua = urb->urb.context; + unsigned long flags; + + if (unlikely(urb->urb.status == -ENOENT || /* unlinked */ + urb->urb.status == -ENODEV || /* device removed */ + urb->urb.status == -ECONNRESET || /* unlinked */ + urb->urb.status == -ESHUTDOWN)) { /* device disabled */ + abort_usb_playback(ua); + abort_alsa_playback(ua); + return; + } + + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states)) { + /* append URB to FIFO */ + spin_lock_irqsave(&ua->lock, flags); + list_add_tail(&urb->ready_list, &ua->ready_playback_urbs); + if (ua->rate_feedback_count > 0) + tasklet_schedule(&ua->playback_tasklet); + ua->playback.substream->runtime->delay -= + urb->urb.iso_frame_desc[0].length / + ua->playback.frame_bytes; + spin_unlock_irqrestore(&ua->lock, flags); + } +} + +static void first_playback_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + + urb->complete = playback_urb_complete; + playback_urb_complete(urb); + + set_bit(PLAYBACK_URB_COMPLETED, &ua->states); + wake_up(&ua->alsa_playback_wait); +} + +/* copy data from the ALSA ring buffer into the URB buffer */ +static bool copy_playback_data(struct ua101_stream *stream, struct urb *urb, + unsigned int frames) +{ + struct snd_pcm_runtime *runtime; + unsigned int frame_bytes, frames1; + const u8 *source; + + runtime = stream->substream->runtime; + frame_bytes = stream->frame_bytes; + source = runtime->dma_area + stream->buffer_pos * frame_bytes; + if (stream->buffer_pos + frames <= runtime->buffer_size) { + memcpy(urb->transfer_buffer, source, frames * frame_bytes); + } else { + /* wrap around at end of ring buffer */ + frames1 = runtime->buffer_size - stream->buffer_pos; + memcpy(urb->transfer_buffer, source, frames1 * frame_bytes); + memcpy(urb->transfer_buffer + frames1 * frame_bytes, + runtime->dma_area, (frames - frames1) * frame_bytes); + } + + stream->buffer_pos += frames; + if (stream->buffer_pos >= runtime->buffer_size) + stream->buffer_pos -= runtime->buffer_size; + stream->period_pos += frames; + if (stream->period_pos >= runtime->period_size) { + stream->period_pos -= runtime->period_size; + return true; + } + return false; +} + +static inline void add_with_wraparound(struct ua101 *ua, + unsigned int *value, unsigned int add) +{ + *value += add; + if (*value >= ua->playback.queue_length) + *value -= ua->playback.queue_length; +} + +static void playback_tasklet(unsigned long data) +{ + struct ua101 *ua = (void *)data; + unsigned long flags; + unsigned int frames; + struct ua101_urb *urb; + bool do_period_elapsed = false; + int err; + + if (unlikely(!test_bit(USB_PLAYBACK_RUNNING, &ua->states))) + return; + + /* + * Synchronizing the playback rate to the capture rate is done by using + * the same sequence of packet sizes for both streams. + * Submitting a playback URB therefore requires both a ready URB and + * the size of the corresponding capture packet, i.e., both playback + * and capture URBs must have been completed. Since the USB core does + * not guarantee that playback and capture complete callbacks are + * called alternately, we use two FIFOs for packet sizes and read URBs; + * submitting playback URBs is possible as long as both FIFOs are + * nonempty. + */ + spin_lock_irqsave(&ua->lock, flags); + while (ua->rate_feedback_count > 0 && + !list_empty(&ua->ready_playback_urbs)) { + /* take packet size out of FIFO */ + frames = ua->rate_feedback[ua->rate_feedback_start]; + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + ua->rate_feedback_count--; + + /* take URB out of FIFO */ + urb = list_first_entry(&ua->ready_playback_urbs, + struct ua101_urb, ready_list); + list_del(&urb->ready_list); + + /* fill packet with data or silence */ + urb->urb.iso_frame_desc[0].length = + frames * ua->playback.frame_bytes; + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + do_period_elapsed |= copy_playback_data(&ua->playback, + &urb->urb, + frames); + else + memset(urb->urb.transfer_buffer, 0, + urb->urb.iso_frame_desc[0].length); + + /* and off you go ... */ + err = usb_submit_urb(&urb->urb, GFP_ATOMIC); + if (unlikely(err < 0)) { + spin_unlock_irqrestore(&ua->lock, flags); + abort_usb_playback(ua); + abort_alsa_playback(ua); + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + return; + } + ua->playback.substream->runtime->delay += frames; + } + spin_unlock_irqrestore(&ua->lock, flags); + if (do_period_elapsed) + snd_pcm_period_elapsed(ua->playback.substream); +} + +/* copy data from the URB buffer into the ALSA ring buffer */ +static bool copy_capture_data(struct ua101_stream *stream, struct urb *urb, + unsigned int frames) +{ + struct snd_pcm_runtime *runtime; + unsigned int frame_bytes, frames1; + u8 *dest; + + runtime = stream->substream->runtime; + frame_bytes = stream->frame_bytes; + dest = runtime->dma_area + stream->buffer_pos * frame_bytes; + if (stream->buffer_pos + frames <= runtime->buffer_size) { + memcpy(dest, urb->transfer_buffer, frames * frame_bytes); + } else { + /* wrap around at end of ring buffer */ + frames1 = runtime->buffer_size - stream->buffer_pos; + memcpy(dest, urb->transfer_buffer, frames1 * frame_bytes); + memcpy(runtime->dma_area, + urb->transfer_buffer + frames1 * frame_bytes, + (frames - frames1) * frame_bytes); + } + + stream->buffer_pos += frames; + if (stream->buffer_pos >= runtime->buffer_size) + stream->buffer_pos -= runtime->buffer_size; + stream->period_pos += frames; + if (stream->period_pos >= runtime->period_size) { + stream->period_pos -= runtime->period_size; + return true; + } + return false; +} + +static void capture_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + struct ua101_stream *stream = &ua->capture; + unsigned long flags; + unsigned int frames, write_ptr; + bool do_period_elapsed; + int err; + + if (unlikely(urb->status == -ENOENT || /* unlinked */ + urb->status == -ENODEV || /* device removed */ + urb->status == -ECONNRESET || /* unlinked */ + urb->status == -ESHUTDOWN)) /* device disabled */ + goto stream_stopped; + + if (urb->status >= 0 && urb->iso_frame_desc[0].status >= 0) + frames = urb->iso_frame_desc[0].actual_length / + stream->frame_bytes; + else + frames = 0; + + spin_lock_irqsave(&ua->lock, flags); + + if (frames > 0 && test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + do_period_elapsed = copy_capture_data(stream, urb, frames); + else + do_period_elapsed = false; + + if (test_bit(USB_CAPTURE_RUNNING, &ua->states)) { + err = usb_submit_urb(urb, GFP_ATOMIC); + if (unlikely(err < 0)) { + spin_unlock_irqrestore(&ua->lock, flags); + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + goto stream_stopped; + } + + /* append packet size to FIFO */ + write_ptr = ua->rate_feedback_start; + add_with_wraparound(ua, &write_ptr, ua->rate_feedback_count); + ua->rate_feedback[write_ptr] = frames; + if (ua->rate_feedback_count < ua->playback.queue_length) { + ua->rate_feedback_count++; + if (ua->rate_feedback_count == + ua->playback.queue_length) + wake_up(&ua->rate_feedback_wait); + } else { + /* + * Ring buffer overflow; this happens when the playback + * stream is not running. Throw away the oldest entry, + * so that the playback stream, when it starts, sees + * the most recent packet sizes. + */ + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + } + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states) && + !list_empty(&ua->ready_playback_urbs)) + tasklet_schedule(&ua->playback_tasklet); + } + + spin_unlock_irqrestore(&ua->lock, flags); + + if (do_period_elapsed) + snd_pcm_period_elapsed(stream->substream); + + /* for debugging: measure the sample rate relative to the USB clock */ + ua->fps[ua->frame_counter++ / ua->packets_per_second] += frames; + if (ua->frame_counter >= ARRAY_SIZE(ua->fps) * ua->packets_per_second) { + printk(KERN_DEBUG "capture rate:"); + for (frames = 0; frames < ARRAY_SIZE(ua->fps); ++frames) + printk(KERN_CONT " %u", ua->fps[frames]); + printk(KERN_CONT "\n"); + memset(ua->fps, 0, sizeof(ua->fps)); + ua->frame_counter = 0; + } + return; + +stream_stopped: + abort_usb_playback(ua); + abort_usb_capture(ua); + abort_alsa_playback(ua); + abort_alsa_capture(ua); +} + +static void first_capture_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + + urb->complete = capture_urb_complete; + capture_urb_complete(urb); + + set_bit(CAPTURE_URB_COMPLETED, &ua->states); + wake_up(&ua->alsa_capture_wait); +} + +static int submit_stream_urbs(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) { + int err = usb_submit_urb(&stream->urbs[i]->urb, GFP_KERNEL); + if (err < 0) { + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + return err; + } + } + return 0; +} + +static void kill_stream_urbs(struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) + usb_kill_urb(&stream->urbs[i]->urb); +} + +static int enable_iso_interface(struct ua101 *ua, unsigned int intf_index) +{ + struct usb_host_interface *alts; + + alts = ua->intf[intf_index]->cur_altsetting; + if (alts->desc.bAlternateSetting != 1) { + int err = usb_set_interface(ua->dev, + alts->desc.bInterfaceNumber, 1); + if (err < 0) { + dev_err(&ua->dev->dev, + "cannot initialize interface; error %d: %s\n", + err, usb_error_string(err)); + return err; + } + } + return 0; +} + +static void disable_iso_interface(struct ua101 *ua, unsigned int intf_index) +{ + struct usb_host_interface *alts; + + alts = ua->intf[intf_index]->cur_altsetting; + if (alts->desc.bAlternateSetting != 0) { + int err = usb_set_interface(ua->dev, + alts->desc.bInterfaceNumber, 0); + if (err < 0 && !test_bit(DISCONNECTED, &ua->states)) + dev_warn(&ua->dev->dev, + "interface reset failed; error %d: %s\n", + err, usb_error_string(err)); + } +} + +static void stop_usb_capture(struct ua101 *ua) +{ + clear_bit(USB_CAPTURE_RUNNING, &ua->states); + + kill_stream_urbs(&ua->capture); + + disable_iso_interface(ua, INTF_CAPTURE); +} + +static int start_usb_capture(struct ua101 *ua) +{ + int err; + + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + + if (test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return 0; + + kill_stream_urbs(&ua->capture); + + err = enable_iso_interface(ua, INTF_CAPTURE); + if (err < 0) + return err; + + clear_bit(CAPTURE_URB_COMPLETED, &ua->states); + ua->capture.urbs[0]->urb.complete = first_capture_urb_complete; + ua->rate_feedback_start = 0; + ua->rate_feedback_count = 0; + + set_bit(USB_CAPTURE_RUNNING, &ua->states); + err = submit_stream_urbs(ua, &ua->capture); + if (err < 0) + stop_usb_capture(ua); + return err; +} + +static void stop_usb_playback(struct ua101 *ua) +{ + clear_bit(USB_PLAYBACK_RUNNING, &ua->states); + + kill_stream_urbs(&ua->playback); + + tasklet_kill(&ua->playback_tasklet); + + disable_iso_interface(ua, INTF_PLAYBACK); +} + +static int start_usb_playback(struct ua101 *ua) +{ + unsigned int i, frames; + struct urb *urb; + int err = 0; + + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return 0; + + kill_stream_urbs(&ua->playback); + tasklet_kill(&ua->playback_tasklet); + + err = enable_iso_interface(ua, INTF_PLAYBACK); + if (err < 0) + return err; + + clear_bit(PLAYBACK_URB_COMPLETED, &ua->states); + ua->playback.urbs[0]->urb.complete = + first_playback_urb_complete; + spin_lock_irq(&ua->lock); + INIT_LIST_HEAD(&ua->ready_playback_urbs); + spin_unlock_irq(&ua->lock); + + /* + * We submit the initial URBs all at once, so we have to wait for the + * packet size FIFO to be full. + */ + wait_event(ua->rate_feedback_wait, + ua->rate_feedback_count >= ua->playback.queue_length || + !test_bit(USB_CAPTURE_RUNNING, &ua->states) || + test_bit(DISCONNECTED, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) { + stop_usb_playback(ua); + return -ENODEV; + } + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) { + stop_usb_playback(ua); + return -EIO; + } + + for (i = 0; i < ua->playback.queue_length; ++i) { + /* all initial URBs contain silence */ + spin_lock_irq(&ua->lock); + frames = ua->rate_feedback[ua->rate_feedback_start]; + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + ua->rate_feedback_count--; + spin_unlock_irq(&ua->lock); + urb = &ua->playback.urbs[i]->urb; + urb->iso_frame_desc[0].length = + frames * ua->playback.frame_bytes; + memset(urb->transfer_buffer, 0, + urb->iso_frame_desc[0].length); + } + + set_bit(USB_PLAYBACK_RUNNING, &ua->states); + err = submit_stream_urbs(ua, &ua->playback); + if (err < 0) + stop_usb_playback(ua); + return err; +} + +static void abort_alsa_capture(struct ua101 *ua) +{ + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); +} + +static void abort_alsa_playback(struct ua101 *ua) +{ + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); +} + +static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, + unsigned int channels) +{ + int err; + + substream->runtime->hw.info = + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_FIFO_IN_FRAMES; + substream->runtime->hw.formats = ua->format_bit; + substream->runtime->hw.rates = snd_pcm_rate_to_rate_bit(ua->rate); + substream->runtime->hw.rate_min = ua->rate; + substream->runtime->hw.rate_max = ua->rate; + substream->runtime->hw.channels_min = channels; + substream->runtime->hw.channels_max = channels; + substream->runtime->hw.buffer_bytes_max = 45000 * 1024; + substream->runtime->hw.period_bytes_min = 1; + substream->runtime->hw.period_bytes_max = UINT_MAX; + substream->runtime->hw.periods_min = 2; + substream->runtime->hw.periods_max = UINT_MAX; + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 1500000 / ua->packets_per_second, + 8192000); + if (err < 0) + return err; + err = snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + return err; +} + +static int capture_pcm_open(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + ua->capture.substream = substream; + err = set_stream_hw(ua, substream, ua->capture.channels); + if (err < 0) + return err; + substream->runtime->hw.fifo_size = + DIV_ROUND_CLOSEST(ua->rate, ua->packets_per_second); + substream->runtime->delay = substream->runtime->hw.fifo_size; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + set_bit(ALSA_CAPTURE_OPEN, &ua->states); + mutex_unlock(&ua->mutex); + return err; +} + +static int playback_pcm_open(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + ua->playback.substream = substream; + err = set_stream_hw(ua, substream, ua->playback.channels); + if (err < 0) + return err; + substream->runtime->hw.fifo_size = + DIV_ROUND_CLOSEST(ua->rate * ua->playback.queue_length, + ua->packets_per_second); + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err < 0) + goto error; + err = start_usb_playback(ua); + if (err < 0) { + if (!test_bit(ALSA_CAPTURE_OPEN, &ua->states)) + stop_usb_capture(ua); + goto error; + } + set_bit(ALSA_PLAYBACK_OPEN, &ua->states); +error: + mutex_unlock(&ua->mutex); + return err; +} + +static int capture_pcm_close(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + + mutex_lock(&ua->mutex); + clear_bit(ALSA_CAPTURE_OPEN, &ua->states); + if (!test_bit(ALSA_PLAYBACK_OPEN, &ua->states)) + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + return 0; +} + +static int playback_pcm_close(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + + mutex_lock(&ua->mutex); + stop_usb_playback(ua); + clear_bit(ALSA_PLAYBACK_OPEN, &ua->states); + if (!test_bit(ALSA_CAPTURE_OPEN, &ua->states)) + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + return 0; +} + +static int capture_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + return snd_pcm_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} + +static int playback_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + err = start_usb_playback(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + return snd_pcm_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} + +static int ua101_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_free_vmalloc_buffer(substream); + return 0; +} + +static int capture_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + /* + * The EHCI driver schedules the first packet of an iso stream at 10 ms + * in the future, i.e., no data is actually captured for that long. + * Take the wait here so that the stream is known to be actually + * running when the start trigger has been called. + */ + wait_event(ua->alsa_capture_wait, + test_bit(CAPTURE_URB_COMPLETED, &ua->states) || + !test_bit(USB_CAPTURE_RUNNING, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return -EIO; + + ua->capture.period_pos = 0; + ua->capture.buffer_pos = 0; + return 0; +} + +static int playback_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + err = start_usb_playback(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + /* see the comment in capture_pcm_prepare() */ + wait_event(ua->alsa_playback_wait, + test_bit(PLAYBACK_URB_COMPLETED, &ua->states) || + !test_bit(USB_PLAYBACK_RUNNING, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + if (!test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return -EIO; + + substream->runtime->delay = 0; + ua->playback.period_pos = 0; + ua->playback.buffer_pos = 0; + return 0; +} + +static int capture_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ua101 *ua = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return -EIO; + set_bit(ALSA_CAPTURE_RUNNING, &ua->states); + return 0; + case SNDRV_PCM_TRIGGER_STOP: + clear_bit(ALSA_CAPTURE_RUNNING, &ua->states); + return 0; + default: + return -EINVAL; + } +} + +static int playback_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ua101 *ua = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (!test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return -EIO; + set_bit(ALSA_PLAYBACK_RUNNING, &ua->states); + return 0; + case SNDRV_PCM_TRIGGER_STOP: + clear_bit(ALSA_PLAYBACK_RUNNING, &ua->states); + return 0; + default: + return -EINVAL; + } +} + +static inline snd_pcm_uframes_t ua101_pcm_pointer(struct ua101 *ua, + struct ua101_stream *stream) +{ + unsigned long flags; + unsigned int pos; + + spin_lock_irqsave(&ua->lock, flags); + pos = stream->buffer_pos; + spin_unlock_irqrestore(&ua->lock, flags); + return pos; +} + +static snd_pcm_uframes_t capture_pcm_pointer(struct snd_pcm_substream *subs) +{ + struct ua101 *ua = subs->private_data; + + return ua101_pcm_pointer(ua, &ua->capture); +} + +static snd_pcm_uframes_t playback_pcm_pointer(struct snd_pcm_substream *subs) +{ + struct ua101 *ua = subs->private_data; + + return ua101_pcm_pointer(ua, &ua->playback); +} + +static struct snd_pcm_ops capture_pcm_ops = { + .open = capture_pcm_open, + .close = capture_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = capture_pcm_hw_params, + .hw_free = ua101_pcm_hw_free, + .prepare = capture_pcm_prepare, + .trigger = capture_pcm_trigger, + .pointer = capture_pcm_pointer, + .page = snd_pcm_get_vmalloc_page, +}; + +static struct snd_pcm_ops playback_pcm_ops = { + .open = playback_pcm_open, + .close = playback_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = playback_pcm_hw_params, + .hw_free = ua101_pcm_hw_free, + .prepare = playback_pcm_prepare, + .trigger = playback_pcm_trigger, + .pointer = playback_pcm_pointer, + .page = snd_pcm_get_vmalloc_page, +}; + +static const struct uac_format_type_i_discrete_descriptor * +find_format_descriptor(struct usb_interface *interface) +{ + struct usb_host_interface *alt; + u8 *extra; + int extralen; + + if (interface->num_altsetting != 2) { + dev_err(&interface->dev, "invalid num_altsetting\n"); + return NULL; + } + + alt = &interface->altsetting[0]; + if (alt->desc.bNumEndpoints != 0) { + dev_err(&interface->dev, "invalid bNumEndpoints\n"); + return NULL; + } + + alt = &interface->altsetting[1]; + if (alt->desc.bNumEndpoints != 1) { + dev_err(&interface->dev, "invalid bNumEndpoints\n"); + return NULL; + } + + extra = alt->extra; + extralen = alt->extralen; + while (extralen >= sizeof(struct usb_descriptor_header)) { + struct uac_format_type_i_discrete_descriptor *desc; + + desc = (struct uac_format_type_i_discrete_descriptor *)extra; + if (desc->bLength > extralen) { + dev_err(&interface->dev, "descriptor overflow\n"); + return NULL; + } + if (desc->bLength == UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(1) && + desc->bDescriptorType == USB_DT_CS_INTERFACE && + desc->bDescriptorSubtype == UAC_FORMAT_TYPE) { + if (desc->bFormatType != UAC_FORMAT_TYPE_I_PCM || + desc->bSamFreqType != 1) { + dev_err(&interface->dev, + "invalid format type\n"); + return NULL; + } + return desc; + } + extralen -= desc->bLength; + extra += desc->bLength; + } + dev_err(&interface->dev, "sample format descriptor not found\n"); + return NULL; +} + +static int detect_usb_format(struct ua101 *ua) +{ + const struct uac_format_type_i_discrete_descriptor *fmt_capture; + const struct uac_format_type_i_discrete_descriptor *fmt_playback; + const struct usb_endpoint_descriptor *epd; + unsigned int rate2; + + fmt_capture = find_format_descriptor(ua->intf[INTF_CAPTURE]); + fmt_playback = find_format_descriptor(ua->intf[INTF_PLAYBACK]); + if (!fmt_capture || !fmt_playback) + return -ENXIO; + + switch (fmt_capture->bSubframeSize) { + case 3: + ua->format_bit = SNDRV_PCM_FMTBIT_S24_3LE; + break; + case 4: + ua->format_bit = SNDRV_PCM_FMTBIT_S32_LE; + break; + default: + dev_err(&ua->dev->dev, "sample width is not 24 or 32 bits\n"); + return -ENXIO; + } + if (fmt_capture->bSubframeSize != fmt_playback->bSubframeSize) { + dev_err(&ua->dev->dev, + "playback/capture sample widths do not match\n"); + return -ENXIO; + } + + if (fmt_capture->bBitResolution != 24 || + fmt_playback->bBitResolution != 24) { + dev_err(&ua->dev->dev, "sample width is not 24 bits\n"); + return -ENXIO; + } + + ua->rate = combine_triple(fmt_capture->tSamFreq[0]); + rate2 = combine_triple(fmt_playback->tSamFreq[0]); + if (ua->rate != rate2) { + dev_err(&ua->dev->dev, + "playback/capture rates do not match: %u/%u\n", + rate2, ua->rate); + return -ENXIO; + } + + switch (ua->dev->speed) { + case USB_SPEED_FULL: + ua->packets_per_second = 1000; + break; + case USB_SPEED_HIGH: + ua->packets_per_second = 8000; + break; + default: + dev_err(&ua->dev->dev, "unknown device speed\n"); + return -ENXIO; + } + + ua->capture.channels = fmt_capture->bNrChannels; + ua->playback.channels = fmt_playback->bNrChannels; + ua->capture.frame_bytes = + fmt_capture->bSubframeSize * ua->capture.channels; + ua->playback.frame_bytes = + fmt_playback->bSubframeSize * ua->playback.channels; + + epd = &ua->intf[INTF_CAPTURE]->altsetting[1].endpoint[0].desc; + if (!usb_endpoint_is_isoc_in(epd)) { + dev_err(&ua->dev->dev, "invalid capture endpoint\n"); + return -ENXIO; + } + ua->capture.usb_pipe = usb_rcvisocpipe(ua->dev, usb_endpoint_num(epd)); + ua->capture.max_packet_bytes = le16_to_cpu(epd->wMaxPacketSize); + + epd = &ua->intf[INTF_PLAYBACK]->altsetting[1].endpoint[0].desc; + if (!usb_endpoint_is_isoc_out(epd)) { + dev_err(&ua->dev->dev, "invalid playback endpoint\n"); + return -ENXIO; + } + ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, usb_endpoint_num(epd)); + ua->playback.max_packet_bytes = le16_to_cpu(epd->wMaxPacketSize); + return 0; +} + +static int alloc_stream_buffers(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int remaining_packets, packets, packets_per_page, i; + size_t size; + + stream->queue_length = queue_length; + stream->queue_length = max(stream->queue_length, + (unsigned int)MIN_QUEUE_LENGTH); + stream->queue_length = min(stream->queue_length, + (unsigned int)MAX_QUEUE_LENGTH); + + /* + * The cache pool sizes used by usb_buffer_alloc() (128, 512, 2048) are + * quite bad when used with the packet sizes of this device (e.g. 280, + * 520, 624). Therefore, we allocate and subdivide entire pages, using + * a smaller buffer only for the last chunk. + */ + remaining_packets = stream->queue_length; + packets_per_page = PAGE_SIZE / stream->max_packet_bytes; + for (i = 0; i < ARRAY_SIZE(stream->buffers); ++i) { + packets = min(remaining_packets, packets_per_page); + size = packets * stream->max_packet_bytes; + stream->buffers[i].addr = + usb_buffer_alloc(ua->dev, size, GFP_KERNEL, + &stream->buffers[i].dma); + if (!stream->buffers[i].addr) + return -ENOMEM; + stream->buffers[i].size = size; + remaining_packets -= packets; + if (!remaining_packets) + break; + } + if (remaining_packets) { + dev_err(&ua->dev->dev, "too many packets\n"); + return -ENXIO; + } + return 0; +} + +static void free_stream_buffers(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(stream->buffers); ++i) + usb_buffer_free(ua->dev, + stream->buffers[i].size, + stream->buffers[i].addr, + stream->buffers[i].dma); +} + +static int alloc_stream_urbs(struct ua101 *ua, struct ua101_stream *stream, + void (*urb_complete)(struct urb *)) +{ + unsigned max_packet_size = stream->max_packet_bytes; + struct ua101_urb *urb; + unsigned int b, u = 0; + + for (b = 0; b < ARRAY_SIZE(stream->buffers); ++b) { + unsigned int size = stream->buffers[b].size; + u8 *addr = stream->buffers[b].addr; + dma_addr_t dma = stream->buffers[b].dma; + + while (size >= max_packet_size) { + if (u >= stream->queue_length) + goto bufsize_error; + urb = kmalloc(sizeof(*urb), GFP_KERNEL); + if (!urb) + return -ENOMEM; + usb_init_urb(&urb->urb); + urb->urb.dev = ua->dev; + urb->urb.pipe = stream->usb_pipe; + urb->urb.transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + urb->urb.transfer_buffer = addr; + urb->urb.transfer_dma = dma; + urb->urb.transfer_buffer_length = max_packet_size; + urb->urb.number_of_packets = 1; + urb->urb.interval = 1; + urb->urb.context = ua; + urb->urb.complete = urb_complete; + urb->urb.iso_frame_desc[0].offset = 0; + urb->urb.iso_frame_desc[0].length = max_packet_size; + stream->urbs[u++] = urb; + size -= max_packet_size; + addr += max_packet_size; + dma += max_packet_size; + } + } + if (u == stream->queue_length) + return 0; +bufsize_error: + dev_err(&ua->dev->dev, "internal buffer size error\n"); + return -ENXIO; +} + +static void free_stream_urbs(struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) + kfree(stream->urbs[i]); +} + +static void free_usb_related_resources(struct ua101 *ua, + struct usb_interface *interface) +{ + unsigned int i; + + free_stream_urbs(&ua->capture); + free_stream_urbs(&ua->playback); + free_stream_buffers(ua, &ua->capture); + free_stream_buffers(ua, &ua->playback); + + for (i = 0; i < ARRAY_SIZE(ua->intf); ++i) + if (ua->intf[i]) { + usb_set_intfdata(ua->intf[i], NULL); + if (ua->intf[i] != interface) + usb_driver_release_interface(&ua101_driver, + ua->intf[i]); + } +} + +static void ua101_card_free(struct snd_card *card) +{ + struct ua101 *ua = card->private_data; + + mutex_destroy(&ua->mutex); +} + +static int ua101_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + static const struct snd_usb_midi_endpoint_info midi_ep = { + .out_cables = 0x0001, + .in_cables = 0x0001 + }; + static const struct snd_usb_audio_quirk midi_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &midi_ep + }; + struct snd_card *card; + struct ua101 *ua; + unsigned int card_index, i; + char usb_path[32]; + int err; + + if (interface->altsetting->desc.bInterfaceNumber != 0) + return -ENODEV; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*ua), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + card->private_free = ua101_card_free; + ua = card->private_data; + ua->dev = interface_to_usbdev(interface); + ua->card = card; + ua->card_index = card_index; + INIT_LIST_HEAD(&ua->midi_list); + spin_lock_init(&ua->lock); + mutex_init(&ua->mutex); + INIT_LIST_HEAD(&ua->ready_playback_urbs); + tasklet_init(&ua->playback_tasklet, + playback_tasklet, (unsigned long)ua); + init_waitqueue_head(&ua->alsa_capture_wait); + init_waitqueue_head(&ua->rate_feedback_wait); + init_waitqueue_head(&ua->alsa_playback_wait); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { + ua->intf[2] = interface; + ua->intf[0] = usb_ifnum_to_if(ua->dev, 1); + ua->intf[1] = usb_ifnum_to_if(ua->dev, 2); + usb_driver_claim_interface(&ua101_driver, ua->intf[0], ua); + usb_driver_claim_interface(&ua101_driver, ua->intf[1], ua); + } else { +#endif + ua->intf[0] = interface; + for (i = 1; i < ARRAY_SIZE(ua->intf); ++i) { + ua->intf[i] = usb_ifnum_to_if(ua->dev, i); + if (!ua->intf[i]) { + dev_err(&ua->dev->dev, "interface %u not found\n", i); + err = -ENXIO; + goto probe_error; + } + err = usb_driver_claim_interface(&ua101_driver, + ua->intf[i], ua); + if (err < 0) { + ua->intf[i] = NULL; + err = -EBUSY; + goto probe_error; + } + } +#ifdef UA1A_HACK + } +#endif + + snd_card_set_dev(card, &interface->dev); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { + ua->format_bit = SNDRV_PCM_FMTBIT_S16_LE; + ua->rate = 44100; + ua->packets_per_second = 1000; + ua->capture.channels = 2; + ua->playback.channels = 2; + ua->capture.frame_bytes = 4; + ua->playback.frame_bytes = 4; + ua->capture.usb_pipe = usb_rcvisocpipe(ua->dev, 2); + ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, 1); + ua->capture.max_packet_bytes = 192; + ua->playback.max_packet_bytes = 192; + } else { +#endif + err = detect_usb_format(ua); + if (err < 0) + goto probe_error; +#ifdef UA1A_HACK + } +#endif + + strcpy(card->driver, "UA-101"); + strcpy(card->shortname, "UA-101"); + usb_make_path(ua->dev, usb_path, sizeof(usb_path)); + snprintf(ua->card->longname, sizeof(ua->card->longname), + "EDIROL UA-101 (serial %s), %u Hz at %s, %s speed", + ua->dev->serial ? ua->dev->serial : "?", ua->rate, usb_path, + ua->dev->speed == USB_SPEED_HIGH ? "high" : "full"); + + err = alloc_stream_buffers(ua, &ua->capture); + if (err < 0) + goto probe_error; + err = alloc_stream_buffers(ua, &ua->playback); + if (err < 0) + goto probe_error; + + err = alloc_stream_urbs(ua, &ua->capture, capture_urb_complete); + if (err < 0) + goto probe_error; + err = alloc_stream_urbs(ua, &ua->playback, playback_urb_complete); + if (err < 0) + goto probe_error; + + err = snd_pcm_new(card, "UA-101", 0, 1, 1, &ua->pcm); + if (err < 0) + goto probe_error; + ua->pcm->private_data = ua; + strcpy(ua->pcm->name, "UA-101"); + snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); + snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct != cpu_to_le16(0x0018)) { +#endif + err = snd_usbmidi_create(card, ua->intf[INTF_MIDI], + &ua->midi_list, &midi_quirk); + if (err < 0) + goto probe_error; +#ifdef UA1A_HACK + } +#endif + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, ua); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + free_usb_related_resources(ua, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void ua101_disconnect(struct usb_interface *interface) +{ + struct ua101 *ua = usb_get_intfdata(interface); + struct list_head *midi; + + if (!ua) + return; + + mutex_lock(&devices_mutex); + + set_bit(DISCONNECTED, &ua->states); + wake_up(&ua->rate_feedback_wait); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(ua->card); + + /* make sure that there are no pending USB requests */ + __list_for_each(midi, &ua->midi_list) + snd_usbmidi_disconnect(midi); + abort_alsa_playback(ua); + abort_alsa_capture(ua); + mutex_lock(&ua->mutex); + stop_usb_playback(ua); + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + + free_usb_related_resources(ua, interface); + + devices_used &= ~(1 << ua->card_index); + + snd_card_free_when_closed(ua->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_device_id ua101_ids[] = { +#ifdef UA1A_HACK + { USB_DEVICE(0x0582, 0x0018) }, +#endif + { USB_DEVICE(0x0582, 0x007d) }, + { USB_DEVICE(0x0582, 0x008d) }, + { } +}; +MODULE_DEVICE_TABLE(usb, ua101_ids); + +static struct usb_driver ua101_driver = { + .name = "snd-ua101", + .id_table = ua101_ids, + .probe = ua101_probe, + .disconnect = ua101_disconnect, +#if 0 + .suspend = ua101_suspend, + .resume = ua101_resume, +#endif +}; + +static int __init alsa_card_ua101_init(void) +{ + return usb_register(&ua101_driver); +} + +static void __exit alsa_card_ua101_exit(void) +{ + usb_deregister(&ua101_driver); + mutex_destroy(&devices_mutex); +} + +module_init(alsa_card_ua101_init); +module_exit(alsa_card_ua101_exit); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b074a59..f352141 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3142,59 +3142,6 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, return 0; } -/* - * Create a stream for an Edirol UA-101 interface. - * Copy, paste and modify from Edirol UA-1000 - */ -static int create_ua101_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - static const struct audioformat ua101_format = { - .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - }; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - struct audioformat *fp; - int stream, err; - - if (iface->num_altsetting != 2) - return -ENXIO; - alts = &iface->altsetting[1]; - altsd = get_iface_desc(alts); - if (alts->extralen != 18 || alts->extra[1] != USB_DT_CS_INTERFACE || - altsd->bNumEndpoints != 1) - return -ENXIO; - - fp = kmemdup(&ua101_format, sizeof(*fp), GFP_KERNEL); - if (!fp) - return -ENOMEM; - - fp->channels = alts->extra[11]; - fp->iface = altsd->bInterfaceNumber; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - fp->rate_max = fp->rate_min = combine_triple(&alts->extra[15]); - - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp); - return err; - } - /* FIXME: playback must be synchronized to capture */ - usb_set_interface(chip->dev, fp->iface, 0); - return 0; -} - static int snd_usb_create_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, const struct snd_usb_audio_quirk *quirk); @@ -3406,7 +3353,6 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, - [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 40ba811..9826337 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,7 +159,6 @@ enum quirk_type { QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, - QUIRK_AUDIO_EDIROL_UA101, QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index a892bda..bd6706c 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1266,37 +1266,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -/* Roland UA-101 in High-Speed Mode only */ -{ - USB_DEVICE(0x0582, 0x007d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "UA-101", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA101 - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA101 - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, { /* has ID 0x0081 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0080), -- cgit v0.10.2 From 5b0cb1d850c26893b1468b3a519433a1b7a176be Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 8 Dec 2009 16:13:32 +0100 Subject: ALSA: hda - add more NID->Control mapping This set of changes add missing NID values to some static control elemenents. Also, it handles all "Capture Source" or "Input Source" controls. Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9cfdb77..20100b1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,6 +931,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) #endif list_del(&codec->list); snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); @@ -985,7 +986,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr mutex_init(&codec->control_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 60); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { @@ -1706,7 +1708,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); /** - * snd_hda_ctl-add - Add a control element and assign to the codec + * snd_hda_ctl_add - Add a control element and assign to the codec * @codec: HD-audio codec * @nid: corresponding NID (optional) * @kctl: the control element to assign @@ -1747,6 +1749,35 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_ctl_add); /** + * snd_hda_add_nid - Assign a NID to a control element + * @codec: HD-audio codec + * @nid: corresponding NID (optional) + * @kctl: the control element to assign + * @index: index to kctl + * + * Add the given control element to an array inside the codec instance. + * This function is used when #snd_hda_ctl_add cannot be used for 1:1 + * NID:KCTL mapping - for example "Capture Source" selector. + */ +int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t nid) +{ + struct hda_nid_item *item; + + if (nid > 0) { + item = snd_array_new(&codec->nids); + if (!item) + return -ENOMEM; + item->kctl = kctl; + item->index = index; + item->nid = nid; + return 0; + } + return -EINVAL; +} +EXPORT_SYMBOL_HDA(snd_hda_add_nid); + +/** * snd_hda_ctls_clear - Clear all controls assigned to the given codec * @codec: HD-audio codec */ @@ -1757,6 +1788,7 @@ void snd_hda_ctls_clear(struct hda_codec *codec) for (i = 0; i < codec->mixers.used; i++) snd_ctl_remove(codec->bus->card, items[i].kctl); snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); } /* pseudo device locking @@ -3476,6 +3508,8 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) for (; knew->name; knew++) { struct snd_kcontrol *kctl; + if (knew->iface == -1) /* skip this codec private value */ + continue; kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; @@ -3496,6 +3530,32 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) } EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); +/** + * snd_hda_add_nids - assign nids to controls from the array + * @codec: the HDA codec + * @kctl: struct snd_kcontrol + * @index: index to kctl + * @nids: the array of hda_nid_t + * @size: count of hda_nid_t items + * + * This helper function assigns NIDs in the given array to a control element. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t *nids, unsigned int size) +{ + int err; + + for ( ; size > 0; size--, nids++) { + err = snd_hda_add_nid(codec, kctl, index, *nids); + if (err < 0) + return err; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_add_nids); + #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1d541b7..0d08ad5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -789,6 +789,7 @@ struct hda_codec { u32 *wcaps; struct snd_array mixers; /* list of assigned mixer elements */ + struct snd_array nids; /* list of mapped mixer elements */ struct hda_cache_rec amp_cache; /* cache for amp access */ struct hda_cache_rec cmd_cache; /* cache for other commands */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 092c6a7..5ea2128 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -861,7 +861,8 @@ static int build_input_controls(struct hda_codec *codec) } /* create input MUX if multiple sources are available */ - err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cap_sel, codec)); + err = snd_hda_ctl_add(codec, spec->adc_node->nid, + snd_ctl_new1(&cap_sel, codec)); if (err < 0) return err; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 5778ae8..98cf3f4 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -342,6 +342,8 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); +int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t *nids, unsigned int size); /* * unsolicited event handler @@ -466,11 +468,14 @@ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); struct hda_nid_item { struct snd_kcontrol *kctl; + unsigned int index; hda_nid_t nid; }; int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, struct snd_kcontrol *kctl); +int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t nid); void snd_hda_ctls_clear(struct hda_codec *codec); /* diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c9afc04..2e27d6a 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -61,18 +61,21 @@ static const char *get_wid_type_name(unsigned int wid_value) return "UNKNOWN Widget"; } -static void print_nid_mixers(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) +static void print_nid_array(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid, + struct snd_array *array) { int i; - struct hda_nid_item *items = codec->mixers.list; + struct hda_nid_item *items = array->list, *item; struct snd_kcontrol *kctl; - for (i = 0; i < codec->mixers.used; i++) { - if (items[i].nid == nid) { - kctl = items[i].kctl; + for (i = 0; i < array->used; i++) { + item = &items[i]; + if (item->nid == nid) { + kctl = item->kctl; snd_iprintf(buffer, " Control: name=\"%s\", index=%i, device=%i\n", - kctl->id.name, kctl->id.index, kctl->id.device); + kctl->id.name, kctl->id.index + item->index, + kctl->id.device); } } } @@ -528,7 +531,8 @@ static void print_gpio(struct snd_info_buffer *buffer, (data & (1<mixers); + print_nid_array(buffer, codec, nid, &codec->nids); } static void print_codec_info(struct snd_info_entry *entry, @@ -608,7 +612,8 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " CP"); snd_iprintf(buffer, "\n"); - print_nid_mixers(buffer, codec, nid); + print_nid_array(buffer, codec, nid, &codec->mixers); + print_nid_array(buffer, codec, nid, &codec->nids); print_nid_pcms(buffer, codec, nid); /* volume knob is a special widget that always have connection diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 447eda1..d418842 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -174,6 +174,7 @@ static struct snd_kcontrol_new ad_beep_mixer[] = { static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; + struct snd_kcontrol *kctl; unsigned int i; int err; @@ -239,6 +240,28 @@ static int ad198x_build_controls(struct hda_codec *codec) } ad198x_free_kctls(codec); /* no longer needed */ + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + if (!kctl) + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + + /* assign IEC958 enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, + SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source"); + if (kctl) { + err = snd_hda_add_nid(codec, kctl, 0, + spec->multiout.dig_out_nid); + if (err < 0) + return err; + } + return 0; } @@ -701,6 +724,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "External Amplifier", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = ad198x_eapd_info, .get = ad198x_eapd_get, .put = ad198x_eapd_put, @@ -808,6 +832,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1a, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -1608,6 +1633,7 @@ static struct snd_kcontrol_new ad1981_hp_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x05, .name = "Master Playback Switch", .info = ad198x_eapd_info, .get = ad198x_eapd_get, @@ -2121,6 +2147,7 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "External Amplifier", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x12, .info = ad198x_eapd_info, .get = ad198x_eapd_get, .put = ad198x_eapd_put, @@ -2242,6 +2269,7 @@ static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "IEC958 Playback Source", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = ad1988_spdif_playback_source_info, .get = ad1988_spdif_playback_source_get, .put = ad1988_spdif_playback_source_put, @@ -3728,6 +3756,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3756,6 +3785,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4097,6 +4127,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4b200da..d0b8c6d 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -759,6 +759,10 @@ static int build_input(struct hda_codec *codec) err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; + err = snd_hda_add_nids(codec, kctl, 0, spec->adc_nid, + spec->num_inputs); + if (err < 0) + return err; } if (spec->num_inputs > 1 && !spec->mic_detect) { diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index a45c116..cc1c223 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -315,7 +315,8 @@ static struct hda_verb cmi9880_allout_init[] = { static int cmi9880_build_controls(struct hda_codec *codec) { struct cmi_spec *spec = codec->spec; - int err; + struct snd_kcontrol *kctl; + int i, err; err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer); if (err < 0) @@ -340,6 +341,15 @@ static int cmi9880_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->adc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 888b631..6b0b872 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -627,6 +627,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, #define ALC_PIN_MODE(xname, nid, dir) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_pin_mode_info, \ .get = alc_pin_mode_get, \ .put = alc_pin_mode_put, \ @@ -678,6 +679,7 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, } #define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_gpio_data_info, \ .get = alc_gpio_data_get, \ .put = alc_gpio_data_put, \ @@ -732,6 +734,7 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, } #define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_spdif_ctrl_info, \ .get = alc_spdif_ctrl_get, \ .put = alc_spdif_ctrl_put, \ @@ -785,6 +788,7 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, #define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_eapd_ctrl_info, \ .get = alc_eapd_ctrl_get, \ .put = alc_eapd_ctrl_put, \ @@ -2410,6 +2414,15 @@ static const char *alc_slave_sws[] = { * build control elements */ +#define NID_MAPPING (-1) + +#define SUBDEV_SPEAKER_ (0 << 6) +#define SUBDEV_HP_ (1 << 6) +#define SUBDEV_LINE_ (2 << 6) +#define SUBDEV_SPEAKER(x) (SUBDEV_SPEAKER_ | ((x) & 0x3f)) +#define SUBDEV_HP(x) (SUBDEV_HP_ | ((x) & 0x3f)) +#define SUBDEV_LINE(x) (SUBDEV_LINE_ | ((x) & 0x3f)) + static void alc_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -2424,8 +2437,11 @@ static struct snd_kcontrol_new alc_beep_mixer[] = { static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; - int i; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new *knew; + int i, j, err; + unsigned int u; + hda_nid_t nid; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -2494,6 +2510,73 @@ static int alc_build_controls(struct hda_codec *codec) } alc_free_kctls(codec); /* no longer needed */ + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + if (!kctl) + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + if (spec->cap_mixer) { + const char *kname = kctl ? kctl->id.name : NULL; + for (knew = spec->cap_mixer; knew->name; knew++) { + if (kname && strcmp(knew->name, kname) == 0) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nid(codec, kctl, i, + spec->adc_nids[i]); + if (err < 0) + return err; + } + } + } + + /* other nid->control mapping */ + for (i = 0; i < spec->num_mixers; i++) { + for (knew = spec->mixers[i]; knew->name; knew++) { + if (knew->iface != NID_MAPPING) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + if (kctl == NULL) + continue; + u = knew->subdevice; + for (j = 0; j < 4; j++, u >>= 8) { + nid = u & 0x3f; + if (nid == 0) + continue; + switch (u & 0xc0) { + case SUBDEV_SPEAKER_: + nid = spec->autocfg.speaker_pins[nid]; + break; + case SUBDEV_LINE_: + nid = spec->autocfg.line_out_pins[nid]; + break; + case SUBDEV_HP_: + nid = spec->autocfg.hp_pins[nid]; + break; + default: + continue; + } + err = snd_hda_add_nid(codec, kctl, 0, nid); + if (err < 0) + return err; + } + u = knew->private_value; + for (j = 0; j < 4; j++, u >>= 8) { + nid = u & 0xff; + if (nid == 0) + continue; + err = snd_hda_add_nid(codec, kctl, 0, nid); + if (err < 0) + return err; + } + } + } return 0; } @@ -3781,6 +3864,7 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, #define PIN_CTL_TEST(xname,nid) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_test_pin_ctl_info, \ .get = alc_test_pin_ctl_get, \ .put = alc_test_pin_ctl_put, \ @@ -3790,6 +3874,7 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, #define PIN_SRC_TEST(xname,nid) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_test_pin_src_info, \ .get = alc_test_pin_src_get, \ .put = alc_test_pin_src_put, \ @@ -5080,6 +5165,7 @@ static struct snd_kcontrol_new alc260_hp_output_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x11, .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, @@ -5118,6 +5204,7 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x11, .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, @@ -10188,8 +10275,14 @@ static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, .info = snd_ctl_boolean_mono_info, \ .get = alc262_hp_master_sw_get, \ .put = alc262_hp_master_sw_put, \ + }, \ + { \ + .iface = NID_MAPPING, \ + .name = "Master Playback Switch", \ + .private_value = 0x15 | (0x16 << 8) | (0x1b << 16), \ } + static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -10347,6 +10440,12 @@ static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, .info = snd_ctl_boolean_mono_info, \ .get = alc262_hippo_master_sw_get, \ .put = alc262_hippo_master_sw_put, \ + }, \ + { \ + .iface = NID_MAPPING, \ + .name = "Master Playback Switch", \ + .subdevice = SUBDEV_HP(0) | (SUBDEV_LINE(0) << 8) | \ + (SUBDEV_SPEAKER(0) << 16), \ } static struct snd_kcontrol_new alc262_hippo_mixer[] = { @@ -10820,11 +10919,17 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }, + { + .iface = NID_MAPPING, + .name = "Master Playback Switch", + .private_value = 0x1b, + }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), @@ -10855,6 +10960,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -11009,6 +11115,11 @@ static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { .get = alc_mux_enum_get, .put = alc262_ultra_mux_enum_put, }, + { + .iface = NID_MAPPING, + .name = "Capture Source", + .private_value = 0x15, + }, { } /* end */ }; @@ -12026,6 +12137,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12041,6 +12153,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12058,6 +12171,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13010,6 +13124,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13030,6 +13145,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 43b436c..f419ee8 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -122,6 +122,7 @@ static int si3054_switch_put(struct snd_kcontrol *kcontrol, #define SI3054_KCONTROL(kname,reg,mask) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = kname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | reg, \ .info = si3054_switch_info, \ .get = si3054_switch_get, \ .put = si3054_switch_put, \ diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b70e26a..64995e8 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -54,6 +54,8 @@ #include "hda_codec.h" #include "hda_local.h" +#define NID_MAPPING (-1) + /* amp values */ #define AMP_VAL_IDX_SHIFT 19 #define AMP_VAL_IDX_MASK (0x0f<<19) @@ -157,6 +159,19 @@ struct via_spec { #endif }; +static struct via_spec * via_new_spec(struct hda_codec *codec) +{ + struct via_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return NULL; + + codec->spec = spec; + spec->codec = codec; + return spec; +} + static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { u32 vendor_id = codec->vendor_id; @@ -448,6 +463,22 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, return 0; } +static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec, + struct snd_kcontrol_new *tmpl) +{ + struct snd_kcontrol_new *knew; + + snd_array_init(&spec->kctls, sizeof(*knew), 32); + knew = snd_array_new(&spec->kctls); + if (!knew) + return NULL; + *knew = *tmpl; + knew->name = kstrdup(tmpl->name, GFP_KERNEL); + if (!knew->name) + return NULL; + return 0; +} + static void via_free_kctls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1088,24 +1119,9 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct via_spec *spec = codec->spec; - hda_nid_t nid; + hda_nid_t nid = kcontrol->private_value; unsigned int pinsel; - switch (spec->codec_type) { - case VT1718S: - nid = 0x34; - break; - case VT2002P: - nid = 0x35; - break; - case VT1812: - nid = 0x3d; - break; - default: - nid = spec->autocfg.hp_pins[0]; - break; - } /* use !! to translate conn sel 2 for VT1718S */ pinsel = !!snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, @@ -1127,29 +1143,24 @@ static void activate_ctl(struct hda_codec *codec, const char *name, int active) } } +static hda_nid_t side_mute_channel(struct via_spec *spec) +{ + switch (spec->codec_type) { + case VT1708: return 0x1b; + case VT1709_10CH: return 0x29; + case VT1708B_8CH: /* fall thru */ + case VT1708S: return 0x27; + default: return 0; + } +} + static int update_side_mute_status(struct hda_codec *codec) { /* mute side channel */ struct via_spec *spec = codec->spec; unsigned int parm = spec->hp_independent_mode ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; - hda_nid_t sw3; - - switch (spec->codec_type) { - case VT1708: - sw3 = 0x1b; - break; - case VT1709_10CH: - sw3 = 0x29; - break; - case VT1708B_8CH: - case VT1708S: - sw3 = 0x27; - break; - default: - sw3 = 0; - break; - } + hda_nid_t sw3 = side_mute_channel(spec); if (sw3) snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1162,28 +1173,11 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - hda_nid_t nid = spec->autocfg.hp_pins[0]; + hda_nid_t nid = kcontrol->private_value; unsigned int pinsel = ucontrol->value.enumerated.item[0]; /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - - switch (spec->codec_type) { - case VT1718S: - nid = 0x34; - pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ - spec->multiout.num_dacs = 4; - break; - case VT2002P: - nid = 0x35; - break; - case VT1812: - nid = 0x3d; - break; - default: - nid = spec->autocfg.hp_pins[0]; - break; - } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid @@ -1207,18 +1201,55 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new via_hp_mixer[] = { +static struct snd_kcontrol_new via_hp_mixer[2] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Independent HP", - .count = 1, .info = via_independent_hp_info, .get = via_independent_hp_get, .put = via_independent_hp_put, }, - { } /* end */ + { + .iface = NID_MAPPING, + .name = "Independent HP", + }, }; +static int via_hp_build(struct via_spec *spec) +{ + struct snd_kcontrol_new *knew; + hda_nid_t nid; + + knew = via_clone_control(spec, &via_hp_mixer[0]); + if (knew == NULL) + return -ENOMEM; + + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + break; + case VT2002P: + nid = 0x35; + break; + case VT1812: + nid = 0x3d; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } + + knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; + knew->private_value = nid; + + knew = via_clone_control(spec, &via_hp_mixer[1]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = side_mute_channel(spec); + + return 0; +} + static void notify_aa_path_ctls(struct hda_codec *codec) { int i; @@ -1376,7 +1407,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new via_smart51_mixer[] = { +static struct snd_kcontrol_new via_smart51_mixer[2] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Smart 5.1", @@ -1385,9 +1416,36 @@ static struct snd_kcontrol_new via_smart51_mixer[] = { .get = via_smart51_get, .put = via_smart51_put, }, - {} /* end */ + { + .iface = NID_MAPPING, + .name = "Smart 5.1", + } }; +static int via_smart51_build(struct via_spec *spec) +{ + struct snd_kcontrol_new *knew; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + hda_nid_t nid; + int i; + + knew = via_clone_control(spec, &via_smart51_mixer[0]); + if (knew == NULL) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + knew = via_clone_control(spec, &via_smart51_mixer[1]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = nid; + } + } + + return 0; +} + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -1819,8 +1877,9 @@ static struct hda_pcm_stream vt1708_pcm_digital_capture = { static int via_build_controls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; - int err; - int i; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new *knew; + int err, i; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -1845,6 +1904,28 @@ static int via_build_controls(struct hda_codec *codec) return err; } + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->mux_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + + /* other nid->control mapping */ + for (i = 0; i < spec->num_mixers; i++) { + for (knew = spec->mixers[i]; knew->name; knew++) { + if (knew->iface != NID_MAPPING) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + if (kctl == NULL) + continue; + err = snd_hda_add_nid(codec, kctl, 0, + knew->subdevice); + } + } + /* init power states */ set_jack_power_state(codec); analog_low_current_mode(codec, 1); @@ -2481,9 +2562,9 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -2554,12 +2635,10 @@ static int patch_vt1708(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708_parse_auto_config(codec); if (err < 0) { @@ -2597,7 +2676,6 @@ static int patch_vt1708(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - spec->codec = codec; INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; } @@ -3010,9 +3088,9 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -3032,12 +3110,10 @@ static int patch_vt1709_10ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - err = vt1709_parse_auto_config(codec); if (err < 0) { via_free(codec); @@ -3126,12 +3202,10 @@ static int patch_vt1709_6ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - err = vt1709_parse_auto_config(codec); if (err < 0) { via_free(codec); @@ -3581,9 +3655,9 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -3605,12 +3679,10 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) if (get_codec_type(codec) == VT1708BCE) return patch_vt1708S(codec); /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708B_parse_auto_config(codec); if (err < 0) { @@ -3657,12 +3729,10 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708B_parse_auto_config(codec); if (err < 0) { @@ -4071,9 +4141,9 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -4103,12 +4173,10 @@ static int patch_vt1708S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708S_parse_auto_config(codec); if (err < 0) { @@ -4443,7 +4511,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -4464,12 +4532,10 @@ static int patch_vt1702(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1702_parse_auto_config(codec); if (err < 0) { @@ -4865,9 +4931,9 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -4888,12 +4954,10 @@ static int patch_vt1718S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1718S_parse_auto_config(codec); if (err < 0) { @@ -5014,6 +5078,7 @@ static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Mic Capture Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x26, .count = 1, .info = vt1716s_dmic_info, .get = vt1716s_dmic_get, @@ -5361,9 +5426,9 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -5384,12 +5449,10 @@ static int patch_vt1716S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1716S_parse_auto_config(codec); if (err < 0) { @@ -5719,7 +5782,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -5741,12 +5804,10 @@ static int patch_vt2002P(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt2002P_parse_auto_config(codec); if (err < 0) { @@ -6070,7 +6131,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -6092,12 +6153,10 @@ static int patch_vt1812(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1812_parse_auto_config(codec); if (err < 0) { -- cgit v0.10.2 From 9e3fd8719f624a43575b56a4777b1552399a8be8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 8 Dec 2009 17:45:25 +0100 Subject: ALSA: hda - introduce HDA_SUBDEV_AMP_FLAG (ControlAmp in proc) The purpose of this changeset is to show information about amplifier setting in the codec proc file. Something like: Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 20100b1..c9af15e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1723,19 +1723,22 @@ EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); * * snd_hda_ctl_add() checks the control subdev id field whether * #HDA_SUBDEV_NID_FLAG bit is set. If set (and @nid is zero), the lower - * bits value is taken as the NID to assign. + * bits value is taken as the NID to assign. The #HDA_NID_ITEM_AMP bit + * specifies if kctl->private_value is a HDA amplifier value. */ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, struct snd_kcontrol *kctl) { int err; + unsigned short flags = 0; struct hda_nid_item *item; - if (kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) { - if (nid == 0) - nid = kctl->id.subdevice & 0xffff; + if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) + flags |= HDA_NID_ITEM_AMP; + if ((kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) != 0 && nid == 0) + nid = kctl->id.subdevice & 0xffff; + if (kctl->id.subdevice & 0xf0000000) kctl->id.subdevice = 0; - } err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) return err; @@ -1744,6 +1747,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, return -ENOMEM; item->kctl = kctl; item->nid = nid; + item->flags = flags; return 0; } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 98cf3f4..0a25647 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -31,6 +31,7 @@ * in snd_hda_ctl_add(), so that this value won't appear in the outside. */ #define HDA_SUBDEV_NID_FLAG (1U << 31) +#define HDA_SUBDEV_AMP_FLAG (1U << 30) /* * for mixer controls @@ -42,7 +43,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -63,7 +64,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -81,7 +82,7 @@ /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ @@ -466,10 +467,14 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); +/* flags for hda_nid_item */ +#define HDA_NID_ITEM_AMP (1<<0) + struct hda_nid_item { struct snd_kcontrol *kctl; unsigned int index; hda_nid_t nid; + unsigned short flags; }; int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 2e27d6a..f97d35d 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -76,6 +76,14 @@ static void print_nid_array(struct snd_info_buffer *buffer, " Control: name=\"%s\", index=%i, device=%i\n", kctl->id.name, kctl->id.index + item->index, kctl->id.device); + if (item->flags & HDA_NID_ITEM_AMP) + snd_iprintf(buffer, + " ControlAmp: chs=%lu, dir=%s, " + "idx=%lu, ofs=%lu\n", + get_amp_channels(kctl), + get_amp_direction(kctl) ? "Out" : "In", + get_amp_index(kctl), + get_amp_offset(kctl)); } } } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d418842..5e2bb18 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -832,7 +832,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1a, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1a, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -2602,7 +2602,9 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, if (! knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } @@ -3756,7 +3758,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3785,7 +3787,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4127,7 +4129,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d0b8c6d..e51f665 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -500,6 +500,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); + (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } @@ -513,6 +514,7 @@ static int add_volume(struct hda_codec *codec, const char *name, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); + (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a09c03c..b68650a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2178,6 +2178,7 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x10, .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = snd_hda_mixer_amp_volume_put, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6b0b872..87bf7bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4414,7 +4414,9 @@ static int add_control(struct alc_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } @@ -10919,7 +10921,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, @@ -10960,7 +10962,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1b, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -12137,7 +12139,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12153,7 +12155,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12171,7 +12173,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13124,7 +13126,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13145,7 +13147,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3d59f83..1ee586b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2702,7 +2702,8 @@ stac_control_new(struct sigmatel_spec *spec, return NULL; } if (nid) - knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | nid; return knew; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 64995e8..b94cdee 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -458,7 +458,9 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } -- cgit v0.10.2 From 5e26dfd0615868872cb44842f1e1428c7b414ab0 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 10 Dec 2009 13:57:01 +0100 Subject: ALSA: hda - simplify usage of HDA_SUBDEV_AMP_FLAG The HDA_SUBDEV_NID_FLAG is duplicate for amplifier control elements. Move get_amp_nid_() call to the snd_hda_ctl_add() function. Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c9af15e..c848ec0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1733,11 +1733,14 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, unsigned short flags = 0; struct hda_nid_item *item; - if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) + if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) { flags |= HDA_NID_ITEM_AMP; + if (nid == 0) + nid = get_amp_nid_(kctl->private_value); + } if ((kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) != 0 && nid == 0) nid = kctl->id.subdevice & 0xffff; - if (kctl->id.subdevice & 0xf0000000) + if (kctl->id.subdevice & (HDA_SUBDEV_NID_FLAG|HDA_SUBDEV_AMP_FLAG)) kctl->id.subdevice = 0; err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 0a25647..d505d05 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -43,7 +43,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -64,7 +64,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -82,7 +82,7 @@ /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 5e2bb18..e75b5e5 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -209,9 +209,7 @@ static int ad198x_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, - get_amp_nid_(spec->beep_amp), - kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } @@ -832,7 +830,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1a, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -2602,9 +2600,7 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, if (! knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } @@ -3758,7 +3754,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3787,7 +3783,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4129,7 +4125,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index e51f665..eeb91f6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -501,7 +501,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index, snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; - return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); + return snd_hda_ctl_add(codec, 0, *kctlp); } static int add_volume(struct hda_codec *codec, const char *name, @@ -515,7 +515,7 @@ static int add_volume(struct hda_codec *codec, const char *name, snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; - return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); + return snd_hda_ctl_add(codec, 0, *kctlp); } static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b68650a..1ab2958 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2178,7 +2178,7 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x10, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = snd_hda_mixer_amp_volume_put, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 87bf7bd..cb76795 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2482,8 +2482,7 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, - get_amp_nid_(spec->beep_amp), kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } @@ -4414,9 +4413,7 @@ static int add_control(struct alc_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } @@ -10921,7 +10918,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, @@ -10962,7 +10959,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1b, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -12139,7 +12136,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12155,7 +12152,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12173,7 +12170,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13126,7 +13123,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13147,7 +13144,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1ee586b..0bafea9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2685,7 +2685,7 @@ static struct snd_kcontrol_new * stac_control_new(struct sigmatel_spec *spec, struct snd_kcontrol_new *ktemp, const char *name, - hda_nid_t nid) + unsigned int subdev) { struct snd_kcontrol_new *knew; @@ -2701,9 +2701,7 @@ stac_control_new(struct sigmatel_spec *spec, spec->kctls.alloced--; return NULL; } - if (nid) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | nid; + knew->subdevice = subdev; return knew; } @@ -2713,7 +2711,7 @@ static int stac92xx_add_control_temp(struct sigmatel_spec *spec, unsigned long val) { struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name, - get_amp_nid_(val)); + HDA_SUBDEV_AMP_FLAG); if (!knew) return -ENOMEM; knew->index = idx; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b94cdee..de4839e 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -458,9 +458,7 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } -- cgit v0.10.2 From 926a01ce1ef5e27281af0270e4476979c0522954 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Dec 2009 16:15:00 +0100 Subject: ALSA: Release v1.0.22 Signed-off-by: Jaroslav Kysela diff --git a/include/sound/version.h b/include/sound/version.h index 2293914..1f5d4872 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.21" +#define CONFIG_SND_VERSION "1.0.22" #define CONFIG_SND_DATE "" -- cgit v0.10.2 From 283375cefbf4f91ce51d93d010634c48d0d39044 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Dec 2009 18:09:03 +0000 Subject: ASoC: Push registers out of mixer power decision No need for the mixers to know about this, and it allows for virtual controls. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 846678a..4cf5891 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1262,8 +1262,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, /* test and update the power status of a mixer or switch widget */ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int reg, - int val_mask, int val, int invert) + struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; int found = 0; @@ -1273,9 +1272,6 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, widget->id != snd_soc_dapm_switch) return -ENODEV; - if (!snd_soc_test_bits(widget->codec, reg, val_mask, val)) - return 0; - /* find dapm widget path assoc with kcontrol */ list_for_each_entry(path, &widget->codec->dapm_paths, list) { if (path->kcontrol != kcontrol) @@ -1283,12 +1279,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; - if (val) - /* new connection */ - path->connect = invert ? 0:1; - else - /* old connection must be powered down */ - path->connect = invert ? 1:0; + path->connect = connect; break; } @@ -1695,6 +1686,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val2, val_mask; + int connect; int ret; val = (ucontrol->value.integer.value[0] & mask); @@ -1721,7 +1713,17 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, return 1; } - dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert); + if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { + if (val) + /* new connection */ + connect = invert ? 0:1; + else + /* old connection must be powered down */ + connect = invert ? 1:0; + + dapm_mixer_update_power(widget, kcontrol, connect); + } + if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, kcontrol, -- cgit v0.10.2 From d207c68dd92455a3d618c37b5a9f0dc598723fd6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Dec 2009 17:13:55 +0000 Subject: ASoC: Sort DAPM sequences by CODEC as well In preparation for multiple device support. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4cf5891..de22c2f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -739,6 +739,8 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, struct snd_soc_dapm_widget *b, int sort[]) { + if (a->codec != b->codec) + return (unsigned long)a - (unsigned long)b; if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; if (a->reg != b->reg) -- cgit v0.10.2 From cce2e9db718d823f33ac846c019763cdc84e8658 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Dec 2009 21:50:01 +0000 Subject: ASoC: Register the CODEC in WM8727 Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index d8ffbd6..63a254e 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -44,23 +44,16 @@ struct snd_soc_dai wm8727_dai = { }; EXPORT_SYMBOL_GPL(wm8727_dai); +static struct snd_soc_codec *wm8727_codec; + static int wm8727_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec; int ret = 0; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - mutex_init(&codec->mutex); - codec->name = "WM8727"; - codec->owner = THIS_MODULE; - codec->dai = &wm8727_dai; - codec->num_dai = 1; - socdev->card->codec = codec; - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); + BUG_ON(!wm8727_codec); + + socdev->card->codec = wm8727_codec; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); @@ -80,12 +73,9 @@ pcm_err: static int wm8727_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - if (codec == NULL) - return 0; snd_soc_free_pcms(socdev); - kfree(codec); + return 0; } @@ -98,13 +88,55 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8727); static __devinit int wm8727_platform_probe(struct platform_device *pdev) { + struct snd_soc_codec *codec; + int ret; + + if (wm8727_codec) { + dev_err(&pdev->dev, "Another WM8727 is registered\n"); + return -EBUSY; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + wm8727_codec = codec; + + platform_set_drvdata(pdev, codec); + + mutex_init(&codec->mutex); + codec->dev = &pdev->dev; + codec->name = "WM8727"; + codec->owner = THIS_MODULE; + codec->dai = &wm8727_dai; + codec->num_dai = 1; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + wm8727_dai.dev = &pdev->dev; - return snd_soc_register_dai(&wm8727_dai); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to register CODEC: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8727_dai); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(codec); + return ret; } static int __devexit wm8727_platform_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&wm8727_dai); + snd_soc_unregister_codec(platform_get_drvdata(pdev)); return 0; } -- cgit v0.10.2 From 168db50d967e09133feda8247d4dcb3c73437766 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 9 Dec 2009 13:29:20 +0900 Subject: ASoC: S3C64XX: Remove unnecessary header includes Removed redundant header includes which make no difference to compilation. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index cc7edb5..8feb029 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -15,16 +15,10 @@ #include #include #include -#include #include -#include #include #include -#include -#include -#include -#include #include #include -- cgit v0.10.2 From 0fe692292a26f57b6522fe859cc8b2549ec0cd97 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 9 Dec 2009 13:29:25 +0900 Subject: ASoC: S3C64XX: Compress and generalize the CPU driver The driver can be 'generalized' a bit by not hardcoding '2'(the number of I2Sv3 controllers that the driver can handle) at many places, instead we define a macro for it. That makes it easier to increase number of controllers by changing the parameter at just one place, this will be useful when there is support for newer SoCs, which have the same controller, only more in number. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 8feb029..93ed3aa 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -32,6 +32,11 @@ #include "s3c-dma.h" #include "s3c64xx-i2s.h" +/* The value should be set to maximum of the total number + * of I2Sv3 controllers that any supported SoC has. + */ +#define MAX_I2SV3 2 + static struct s3c2410_dma_client s3c64xx_dma_client_out = { .name = "I2S PCM Stereo out" }; @@ -40,37 +45,12 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { - [0] = { - .channel = DMACH_I2S0_OUT, - .client = &s3c64xx_dma_client_out, - .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD, - .dma_size = 4, - }, - [1] = { - .channel = DMACH_I2S1_OUT, - .client = &s3c64xx_dma_client_out, - .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD, - .dma_size = 4, - }, -}; - -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { - [0] = { - .channel = DMACH_I2S0_IN, - .client = &s3c64xx_dma_client_in, - .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD, - .dma_size = 4, - }, - [1] = { - .channel = DMACH_I2S1_IN, - .client = &s3c64xx_dma_client_in, - .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD, - .dma_size = 4, - }, -}; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; +static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; -static struct s3c_i2sv2_info s3c64xx_i2s[2]; +struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) { @@ -163,55 +143,13 @@ static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { .set_sysclk = s3c64xx_i2s_set_sysclk, }; -struct snd_soc_dai s3c64xx_i2s_dai[] = { - { - .name = "s3c64xx-i2s", - .id = 0, - .probe = s3c64xx_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .ops = &s3c64xx_i2s_dai_ops, - .symmetric_rates = 1, - }, - { - .name = "s3c64xx-i2s", - .id = 1, - .probe = s3c64xx_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, - }, - .ops = &s3c64xx_i2s_dai_ops, - .symmetric_rates = 1, - }, -}; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); - static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) { struct s3c_i2sv2_info *i2s; struct snd_soc_dai *dai; int ret; - if (pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) { + if (pdev->id >= MAX_I2SV3) { dev_err(&pdev->dev, "id %d out of range\n", pdev->id); return -EINVAL; } @@ -219,10 +157,40 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) i2s = &s3c64xx_i2s[pdev->id]; dai = &s3c64xx_i2s_dai[pdev->id]; dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s"; + dai->id = pdev->id; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2s_probe; + dai->ops = &s3c64xx_i2s_dai_ops; i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + if (pdev->id == 0) { + i2s->dma_capture->channel = DMACH_I2S0_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S0_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; + } else { + i2s->dma_capture->channel = DMACH_I2S1_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S1_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; + } + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); if (IS_ERR(i2s->iis_cclk)) { dev_err(&pdev->dev, "failed to get audio-bus\n"); -- cgit v0.10.2 From 7c4e6492205b677a5786b85bcf72ce7c8f4adf15 Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Wed, 9 Dec 2009 12:05:50 +0200 Subject: ASoC: tpa6130a2: Add support for regulator framework Take the regulator framework in use for managing the power sources Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Acked-by: Eduardo Valentin Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 6b650c1..0eb33d4 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -34,10 +35,17 @@ static struct i2c_client *tpa6130a2_client; +#define TPA6130A2_NUM_SUPPLIES 2 +static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { + "CPVSS", + "Vdd", +}; + /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; unsigned char regs[TPA6130A2_CACHEREGNUM]; + struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; int power_gpio; unsigned char power_state; }; @@ -106,10 +114,11 @@ static void tpa6130a2_initialize(void) tpa6130a2_i2c_write(i, data->regs[i]); } -static void tpa6130a2_power(int power) +static int tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; + int ret; BUG_ON(tpa6130a2_client == NULL); data = i2c_get_clientdata(tpa6130a2_client); @@ -117,11 +126,20 @@ static void tpa6130a2_power(int power) mutex_lock(&data->mutex); if (power) { /* Power on */ - if (data->power_gpio >= 0) { + if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); - data->power_state = 1; - tpa6130a2_initialize(); + + ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(&tpa6130a2_client->dev, + "Failed to enable supplies: %d\n", ret); + goto exit; } + + data->power_state = 1; + tpa6130a2_initialize(); + /* Clear SWS */ val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val &= ~TPA6130A2_SWS; @@ -131,13 +149,25 @@ static void tpa6130a2_power(int power) val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val |= TPA6130A2_SWS; tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + /* Power off */ - if (data->power_gpio >= 0) { + if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - data->power_state = 0; + + ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(&tpa6130a2_client->dev, + "Failed to disable supplies: %d\n", ret); + goto exit; } + + data->power_state = 0; } + +exit: mutex_unlock(&data->mutex); + return ret; } static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, @@ -299,15 +329,17 @@ static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w, static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + int ret = 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: - tpa6130a2_power(1); + ret = tpa6130a2_power(1); break; case SND_SOC_DAPM_POST_PMD: - tpa6130a2_power(0); + ret = tpa6130a2_power(0); break; } - return 0; + return ret; } static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { @@ -352,7 +384,7 @@ static int tpa6130a2_probe(struct i2c_client *client, struct device *dev; struct tpa6130a2_data *data; struct tpa6130a2_platform_data *pdata; - int ret; + int i, ret; dev = &client->dev; @@ -387,15 +419,25 @@ static int tpa6130a2_probe(struct i2c_client *client, if (ret < 0) { dev_err(dev, "Failed to request power GPIO (%d)\n", data->power_gpio); - goto fail; + goto err_gpio; } gpio_direction_output(data->power_gpio, 0); - } else { - data->power_state = 1; - tpa6130a2_initialize(); } - tpa6130a2_power(1); + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + + ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + goto err_regulator; + } + + ret = tpa6130a2_power(1); + if (ret != 0) + goto err_power; + /* Read version */ ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) & @@ -404,10 +446,18 @@ static int tpa6130a2_probe(struct i2c_client *client, dev_warn(dev, "UNTESTED version detected (%d)\n", ret); /* Disable the chip */ - tpa6130a2_power(0); + ret = tpa6130a2_power(0); + if (ret != 0) + goto err_power; return 0; -fail: + +err_power: + regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); +err_regulator: + if (data->power_gpio >= 0) + gpio_free(data->power_gpio); +err_gpio: kfree(data); i2c_set_clientdata(tpa6130a2_client, NULL); tpa6130a2_client = NULL; @@ -423,6 +473,9 @@ static int tpa6130a2_remove(struct i2c_client *client) if (data->power_gpio >= 0) gpio_free(data->power_gpio); + + regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + kfree(data); tpa6130a2_client = NULL; -- cgit v0.10.2 From 98615454f66175e923f239ab1d1bd85cd618363e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 14 Dec 2009 13:21:56 +0900 Subject: ASoC: Add DA7210 codec device support for ALSA This original driver was created by Dialog Semiconductor, and cleanuped by Kuninori Morimoto. Special thanks to David Chen. This became very simple ASoC codec driver, and it is tested by EcoVec24 board. Signed-off-by: David Chen Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 011d3ab..691abe7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -23,6 +23,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_DA7210 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -113,6 +114,9 @@ config SND_SOC_AK4671 config SND_SOC_CS4270 tristate +config SND_SOC_DA7210 + tristate + # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0471d90..b328f29 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -10,6 +10,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o +snd-soc-da7210-objs := da7210.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o @@ -67,6 +68,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o +obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c new file mode 100644 index 0000000..14f5f34 --- /dev/null +++ b/sound/soc/codecs/da7210.c @@ -0,0 +1,586 @@ +/* + * DA7210 ALSA Soc codec driver + * + * Copyright (c) 2009 Dialog Semiconductor + * Written by David Chen + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Cleanups by Kuninori Morimoto + * + * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "da7210.h" + +/* DA7210 register space */ +#define DA7210_STATUS 0x02 +#define DA7210_STARTUP1 0x03 +#define DA7210_MIC_L 0x07 +#define DA7210_MIC_R 0x08 +#define DA7210_INMIX_L 0x0D +#define DA7210_INMIX_R 0x0E +#define DA7210_ADC_HPF 0x0F +#define DA7210_ADC 0x10 +#define DA7210_DAC_HPF 0x14 +#define DA7210_DAC_L 0x15 +#define DA7210_DAC_R 0x16 +#define DA7210_DAC_SEL 0x17 +#define DA7210_OUTMIX_L 0x1C +#define DA7210_OUTMIX_R 0x1D +#define DA7210_HP_L_VOL 0x21 +#define DA7210_HP_R_VOL 0x22 +#define DA7210_HP_CFG 0x23 +#define DA7210_DAI_SRC_SEL 0x25 +#define DA7210_DAI_CFG1 0x26 +#define DA7210_DAI_CFG3 0x28 +#define DA7210_PLL_DIV3 0x2B +#define DA7210_PLL 0x2C + +/* STARTUP1 bit fields */ +#define DA7210_SC_MST_EN (1 << 0) + +/* MIC_L bit fields */ +#define DA7210_MICBIAS_EN (1 << 6) +#define DA7210_MIC_L_EN (1 << 7) + +/* MIC_R bit fields */ +#define DA7210_MIC_R_EN (1 << 7) + +/* INMIX_L bit fields */ +#define DA7210_IN_L_EN (1 << 7) + +/* INMIX_R bit fields */ +#define DA7210_IN_R_EN (1 << 7) + +/* ADC_HPF bit fields */ +#define DA7210_ADC_VOICE_EN (1 << 7) + +/* ADC bit fields */ +#define DA7210_ADC_L_EN (1 << 3) +#define DA7210_ADC_R_EN (1 << 7) + +/* DAC_SEL bit fields */ +#define DA7210_DAC_L_SRC_DAI_L (4 << 0) +#define DA7210_DAC_L_EN (1 << 3) +#define DA7210_DAC_R_SRC_DAI_R (5 << 4) +#define DA7210_DAC_R_EN (1 << 7) + +/* OUTMIX_L bit fields */ +#define DA7210_OUT_L_EN (1 << 7) + +/* OUTMIX_R bit fields */ +#define DA7210_OUT_R_EN (1 << 7) + +/* HP_CFG bit fields */ +#define DA7210_HP_2CAP_MODE (1 << 1) +#define DA7210_HP_SENSE_EN (1 << 2) +#define DA7210_HP_L_EN (1 << 3) +#define DA7210_HP_MODE (1 << 6) +#define DA7210_HP_R_EN (1 << 7) + +/* DAI_SRC_SEL bit fields */ +#define DA7210_DAI_OUT_L_SRC (6 << 0) +#define DA7210_DAI_OUT_R_SRC (7 << 4) + +/* DAI_CFG1 bit fields */ +#define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_MASTER (1 << 7) + +/* DAI_CFG3 bit fields */ +#define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_OE (1 << 3) +#define DA7210_DAI_EN (1 << 7) + +/*PLL_DIV3 bit fields */ +#define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) +#define DA7210_PLL_BYP (1 << 6) + +/* PLL bit fields */ +#define DA7210_PLL_FS_48000 (11 << 0) + +#define DA7210_VERSION "0.0.1" + +/* Codec private data */ +struct da7210_priv { + struct snd_soc_codec codec; +}; + +static struct snd_soc_codec *da7210_codec; + +/* + * Register cache + */ +static const u8 da7210_reg[] = { + 0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */ + 0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */ + 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */ + 0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */ + 0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */ + 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */ + 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */ + 0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */ + 0x00, /* R88 */ +}; + +/* + * Read da7210 register cache + */ +static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) +{ + u8 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(da7210_reg)); + return cache[reg]; +} + +/* + * Write to the da7210 register space + */ +static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) +{ + u8 *cache = codec->reg_cache; + u8 data[2]; + + BUG_ON(codec->volatile_register); + + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (reg >= codec->reg_cache_size) + return -EIO; + + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + + cache[reg] = value; + return 0; +} + +/* + * Read from the da7210 register space. + */ +static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) +{ + if (DA7210_STATUS == reg) + return i2c_smbus_read_byte_data(codec->control_data, reg); + + return da7210_read_reg_cache(codec, reg); +} + +static int da7210_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* PlayBack Volume 40 */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + + /* Enable Out */ + snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); + snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); + + } else { + /* Volume 7 */ + snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); + snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); + + /* Enable Mic */ + snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); + snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + } + + return 0; +} + +/* + * Set PCM DAI word length. + */ +static int da7210_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u32 dai_cfg1; + u32 reg, mask; + + /* set DAI source to Left and Right ADC */ + da7210_write(codec, DA7210_DAI_SRC_SEL, + DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); + + /* Enable DAI */ + da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); + + dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + dai_cfg1 |= DA7210_DAI_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dai_cfg1 |= DA7210_DAI_WORD_S24_LE; + break; + default: + return -EINVAL; + } + + da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + + /* FIXME + * + * It support 48K only now + */ + switch (params_rate(params)) { + case 48000: + if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { + reg = DA7210_DAC_HPF; + mask = DA7210_DAC_VOICE_EN; + } else { + reg = DA7210_ADC_HPF; + mask = DA7210_ADC_VOICE_EN; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg, mask, 0); + + return 0; +} + +/* + * Set DAI mode and Format + */ +static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u32 dai_cfg1; + u32 dai_cfg3; + + dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dai_cfg1 |= DA7210_DAI_MODE_MASTER; + break; + default: + return -EINVAL; + } + + /* FIXME + * + * It support I2S only now + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; + break; + default: + return -EINVAL; + } + + /* FIXME + * + * It support 64bit data transmission only now + */ + dai_cfg1 |= DA7210_DAI_FLEN_64BIT; + + da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); + da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + + return 0; +} + +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +/* DAI operations */ +static struct snd_soc_dai_ops da7210_dai_ops = { + .startup = da7210_startup, + .hw_params = da7210_hw_params, + .set_fmt = da7210_set_dai_fmt, +}; + +struct snd_soc_dai da7210_dai = { + .name = "DA7210 IIS", + .id = 0, + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7210_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7210_FORMATS, + }, + .ops = &da7210_dai_ops, +}; +EXPORT_SYMBOL_GPL(da7210_dai); + +/* + * Initialize the DA7210 driver + * register the mixer and dsp interfaces with the kernel + */ +static int da7210_init(struct da7210_priv *da7210) +{ + struct snd_soc_codec *codec = &da7210->codec; + int ret = 0; + + if (da7210_codec) { + dev_err(codec->dev, "Another da7210 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = da7210; + codec->name = "DA7210"; + codec->owner = THIS_MODULE; + codec->read = da7210_read; + codec->write = da7210_write; + codec->dai = &da7210_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache_size = ARRAY_SIZE(da7210_reg); + codec->reg_cache = kmemdup(da7210_reg, + sizeof(da7210_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + da7210_dai.dev = codec->dev; + da7210_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register CODEC: %d\n", ret); + goto init_err; + } + + ret = snd_soc_register_dai(&da7210_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto init_err; + } + + /* FIXME + * + * This driver use fixed value here + */ + + /* + * ADC settings + */ + + /* Enable Left & Right MIC PGA and Mic Bias */ + da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + + /* Enable Left and Right input PGA */ + da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + + /* Enable Left and Right ADC */ + da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); + + /* + * DAC settings + */ + + /* Enable Left and Right DAC */ + da7210_write(codec, DA7210_DAC_SEL, + DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | + DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); + + /* Enable Left and Right out PGA */ + da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + + /* Enable Left and Right HeadPhone PGA */ + da7210_write(codec, DA7210_HP_CFG, + DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | + DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + + /* Diable PLL and bypass it */ + da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); + + /* Bypass PLL and set MCLK freq rang to 10-20MHz */ + da7210_write(codec, DA7210_PLL_DIV3, + DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + + /* Activate all enabled subsystem */ + da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + + return ret; + +init_err: + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + return ret; + +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int da7210_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7210_priv *da7210; + struct snd_soc_codec *codec; + int ret; + + da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + codec = &da7210->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, da7210); + codec->control_data = i2c; + + ret = da7210_init(da7210); + if (ret < 0) + pr_err("Failed to initialise da7210 audio codec\n"); + + return ret; +} + +static int da7210_i2c_remove(struct i2c_client *client) +{ + struct da7210_priv *da7210 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&da7210_dai); + kfree(da7210->codec.reg_cache); + kfree(da7210); + da7210_codec = NULL; + + return 0; +} + +static const struct i2c_device_id da7210_i2c_id[] = { + { "da7210", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da7210_i2c_driver = { + .driver = { + .name = "DA7210 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = da7210_i2c_probe, + .remove = __devexit_p(da7210_i2c_remove), + .id_table = da7210_i2c_id, +}; +#endif + +static int da7210_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (!da7210_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = da7210_codec; + codec = da7210_codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); + +pcm_err: + return ret; +} + +static int da7210_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_da7210 = { + .probe = da7210_probe, + .remove = da7210_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_da7210); + +static int __init da7210_modinit(void) +{ + int ret = 0; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&da7210_i2c_driver); +#endif + return ret; +} +module_init(da7210_modinit); + +static void __exit da7210_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&da7210_i2c_driver); +#endif +} +module_exit(da7210_exit); + +MODULE_DESCRIPTION("ASoC DA7210 driver"); +MODULE_AUTHOR("David Chen, Kuninori Morimoto"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7210.h b/sound/soc/codecs/da7210.h new file mode 100644 index 0000000..390d621 --- /dev/null +++ b/sound/soc/codecs/da7210.h @@ -0,0 +1,24 @@ +/* + * da7210.h -- audio driver for da7210 + * + * Copyright (c) 2009 Dialog Semiconductor + * Written by David Chen + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Cleanups by Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _DA7210_H +#define _DA7210_H + +extern struct snd_soc_dai da7210_dai; +extern struct snd_soc_codec_device soc_codec_dev_da7210; + +#endif + -- cgit v0.10.2 From 038494059f795849012a96adba2ab73e65b94ba5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 14 Dec 2009 13:22:00 +0900 Subject: ASoC: Add FSI-DA7210 sound support for SuperH Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 9e69765..8072a6d 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -47,4 +47,12 @@ config SND_FSI_AK4642 This option enables generic sound support for the FSI - AK4642 unit +config SND_FSI_DA7210 + bool "FSI-DA7210 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_DA7210 + help + This option enables generic sound support for the + FSI - DA7210 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a699787..1d0ec0a 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -13,6 +13,8 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o +snd-soc-fsi-da7210-objs := fsi-da7210.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o +obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c new file mode 100644 index 0000000..33b4d17 --- /dev/null +++ b/sound/soc/sh/fsi-da7210.c @@ -0,0 +1,83 @@ +/* + * fsi-da7210.c + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "../codecs/da7210.h" + +static int fsi_da7210_init(struct snd_soc_codec *codec) +{ + return snd_soc_dai_set_fmt(&da7210_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +} + +static struct snd_soc_dai_link fsi_da7210_dai = { + .name = "DA7210", + .stream_name = "DA7210", + .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .codec_dai = &da7210_dai, + .init = fsi_da7210_init, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_da7210_dai, + .num_links = 1, +}; + +static struct snd_soc_device fsi_da7210_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_da7210, +}; + +static struct platform_device *fsi_da7210_snd_device; + +static int __init fsi_da7210_sound_init(void) +{ + int ret; + + fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_da7210_snd_device) + return -ENOMEM; + + platform_set_drvdata(fsi_da7210_snd_device, &fsi_da7210_snd_devdata); + fsi_da7210_snd_devdata.dev = &fsi_da7210_snd_device->dev; + ret = platform_device_add(fsi_da7210_snd_device); + if (ret) + platform_device_put(fsi_da7210_snd_device); + + return ret; +} + +static void __exit fsi_da7210_sound_exit(void) +{ + platform_device_unregister(fsi_da7210_snd_device); +} + +module_init(fsi_da7210_sound_init); +module_exit(fsi_da7210_sound_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC FSI DA2710"); +MODULE_AUTHOR("Kuninori Morimoto "); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From 3497b91946a3df42830c826939424d98251a3b0d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 15 Dec 2009 20:58:56 +0000 Subject: ASoC: Fix sorting of codecs Makefile entries Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b328f29..c0fd3c8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -96,11 +96,11 @@ obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o -obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o -obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o +obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o -- cgit v0.10.2 From 255173b40db448ce063a2caa680a552fb637ad20 Mon Sep 17 00:00:00 2001 From: Peter Meerwald Date: Mon, 14 Dec 2009 14:44:56 +0100 Subject: ASoC: PLL computation in TLV320AIC3x SoC driver fix precision of PLL computation for TLV320AIC3x SoC driver, test results are at http://pmeerw.net/clk Signed-off-by: Peter Meerwald Acked-by: Vladimir Barinov Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 2b4dc2b..5a8f53c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -765,9 +765,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct aic3x_priv *aic3x = codec->private_data; int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; - u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; - u16 pll_d = 1; + u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u16 d, pll_d = 1; u8 reg; + int clk; /* select data word length */ data = @@ -833,48 +834,70 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, if (bypass_pll) return 0; - /* Use PLL - * find an apropriate setup for j, d, r and p by iterating over - * p and r - j and d are calculated for each fraction. - * Up to 128 values are probed, the closest one wins the game. + /* Use PLL, compute apropriate setup for j, d, r and p, the closest + * one wins the game. Try with d==0 first, next with d!=0. + * Constraints for j are according to the datasheet. * The sysclk is divided by 1000 to prevent integer overflows. */ + codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000); for (r = 1; r <= 16; r++) for (p = 1; p <= 8; p++) { - int clk, tmp = (codec_clk * pll_r * 10) / pll_p; - u8 j = tmp / 10000; - u16 d = tmp % 10000; + for (j = 4; j <= 55; j++) { + /* This is actually 1000*((j+(d/10000))*r)/p + * The term had to be converted to get + * rid of the division by 10000; d = 0 here + */ + int clk = (1000 * j * r) / p; + + /* Check whether this values get closer than + * the best ones we had before + */ + if (abs(codec_clk - clk) < + abs(codec_clk - last_clk)) { + pll_j = j; pll_d = 0; + pll_r = r; pll_p = p; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } + } - if (j > 63) - continue; + /* try with d != 0 */ + for (p = 1; p <= 8; p++) { + j = codec_clk * p / 1000; - if (d != 0 && aic3x->sysclk < 10000000) - continue; + if (j < 4 || j > 11) + continue; - /* This is actually 1000 * ((j + (d/10000)) * r) / p - * The term had to be converted to get rid of the - * division by 10000 */ - clk = ((10000 * j * r) + (d * r)) / (10 * p); + /* do not use codec_clk here since we'd loose precision */ + d = ((2048 * p * fsref) - j * aic3x->sysclk) + * 100 / (aic3x->sysclk/100); - /* check whether this values get closer than the best - * ones we had before */ - if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { - pll_j = j; pll_d = d; pll_r = r; pll_p = p; - last_clk = clk; - } + clk = (10000 * j + d) / (10 * p); - /* Early exit for exact matches */ - if (clk == codec_clk) - break; + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = d; pll_r = 1; pll_p = p; + last_clk = clk; } + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } + if (last_clk == 0) { printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); return -EINVAL; } +found: data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); -- cgit v0.10.2 From c2151433847e88ba05c6bb539d9397ea90d755e6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Dec 2009 20:36:37 +0000 Subject: ASoC: Fix build of DA7210 DAC_VOICE_EN was not defined - looks to have been overly enthusiastically deleted from a previous revision of the patch, pull the value from v1. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 14f5f34..fbf3ab4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -81,6 +81,9 @@ #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) +/* DAC_HPF fields */ +#define DA7210_DAC_VOICE_EN (1 << 7) + /* DAC_SEL bit fields */ #define DA7210_DAC_L_SRC_DAI_L (4 << 0) #define DA7210_DAC_L_EN (1 << 3) -- cgit v0.10.2 From 6c941c8556dd9269be621cd8159fc60e955a91b3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Dec 2009 16:15:00 +0100 Subject: ALSA: Release v1.0.22 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai diff --git a/include/sound/version.h b/include/sound/version.h index 2293914..1f5d4872 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.21" +#define CONFIG_SND_VERSION "1.0.22" #define CONFIG_SND_DATE "" -- cgit v0.10.2 From d1409ae4cecb4af260759bdfdf88fafca23a9940 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Dec 2009 15:01:31 +0100 Subject: ALSA: hda - Fix NULL dereference in kctl-NID mapping in patch_realtek.c capsrc_nids can be NULL, and adc_nids should be taken as fallback. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 36556b1..0124352 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2517,7 +2517,10 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + hda_nid_t *nids = spec->capsrc_nids; + if (!nids) + nids = spec->adc_nids; + err = snd_hda_add_nids(codec, kctl, i, nids, spec->input_mux->num_items); if (err < 0) return err; -- cgit v0.10.2 From 681b84e17747e1c208e8e1acc54cc5e612da84d1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:29:00 +0100 Subject: sound: pcm: add vmalloc buffer helper functions There are now five copies of the code to allocate a PCM buffer using vmalloc(). Add a sixth in the core so that the others can be removed. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a7..0ad2d28 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -905,6 +905,44 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, + size_t size, gfp_t gfp_flags); +int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream); +struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, + unsigned long offset); +#if 0 /* for kernel-doc */ +/** + * snd_pcm_lib_alloc_vmalloc_buffer - allocate virtual DMA buffer + * @substream: the substream to allocate the buffer to + * @size: the requested buffer size, in bytes + * + * Allocates the PCM substream buffer using vmalloc(), i.e., the memory is + * contiguous in kernel virtual space, but not in physical memory. Use this + * if the buffer is accessed by kernel code but not by device DMA. + * + * Returns 1 if the buffer was changed, 0 if not changed, or a negative error + * code. + */ +static int snd_pcm_lib_alloc_vmalloc_buffer + (struct snd_pcm_substream *substream, size_t size); +/** + * snd_pcm_lib_alloc_vmalloc_32_buffer - allocate 32-bit-addressable buffer + * @substream: the substream to allocate the buffer to + * @size: the requested buffer size, in bytes + * + * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses + * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory. + */ +static int snd_pcm_lib_alloc_vmalloc_32_buffer + (struct snd_pcm_substream *substream, size_t size); +#endif +#define snd_pcm_lib_alloc_vmalloc_buffer(subs, size) \ + _snd_pcm_lib_alloc_vmalloc_buffer \ + (subs, size, GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO) +#define snd_pcm_lib_alloc_vmalloc_32_buffer(subs, size) \ + _snd_pcm_lib_alloc_vmalloc_buffer \ + (subs, size, GFP_KERNEL | GFP_DMA32 | __GFP_ZERO) + #ifdef CONFIG_SND_DMA_SGBUF /* * SG-buffer handling diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index caa7796..d9727c7 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -434,3 +434,57 @@ int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(snd_pcm_lib_free_pages); + +int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, + size_t size, gfp_t gfp_flags) +{ + struct snd_pcm_runtime *runtime; + + if (PCM_RUNTIME_CHECK(substream)) + return -EINVAL; + runtime = substream->runtime; + if (runtime->dma_area) { + if (runtime->dma_bytes >= size) + return 0; /* already large enough */ + vfree(runtime->dma_area); + } + runtime->dma_area = __vmalloc(size, gfp_flags, PAGE_KERNEL); + if (!runtime->dma_area) + return -ENOMEM; + runtime->dma_bytes = size; + return 1; +} +EXPORT_SYMBOL(_snd_pcm_lib_alloc_vmalloc_buffer); + +/** + * snd_pcm_lib_free_vmalloc_buffer - free vmalloc buffer + * @substream: the substream with a buffer allocated by + * snd_pcm_lib_alloc_vmalloc_buffer() + */ +int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (PCM_RUNTIME_CHECK(substream)) + return -EINVAL; + runtime = substream->runtime; + vfree(runtime->dma_area); + runtime->dma_area = NULL; + return 0; +} +EXPORT_SYMBOL(snd_pcm_lib_free_vmalloc_buffer); + +/** + * snd_pcm_lib_get_vmalloc_page - map vmalloc buffer offset to page struct + * @substream: the substream with a buffer allocated by + * snd_pcm_lib_alloc_vmalloc_buffer() + * @offset: offset in the buffer + * + * This function is to be used as the page callback in the PCM ops. + */ +struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} +EXPORT_SYMBOL(snd_pcm_lib_get_vmalloc_page); -- cgit v0.10.2 From d20fb5dc076a4cf0fdd00ab5a4e752ea3611e484 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:29:49 +0100 Subject: sound: pdaudiocf: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 5cfa608..0afa683 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -21,7 +21,6 @@ */ #include -#include #include #include #include @@ -29,49 +28,6 @@ /* - * we use a vmalloc'ed (sg-)buffer - */ - -/* get the physical page pointer on the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* - * hw_params callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already enough large */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_32_user(size); - if (! runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* - * hw_free callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - -/* * clear the SRAM contents */ static int pdacf_pcm_clear_sram(struct snd_pdacf *chip) @@ -147,7 +103,8 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd) static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_32_buffer + (subs, params_buffer_bytes(hw_params)); } /* @@ -155,7 +112,7 @@ static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs, */ static int pdacf_pcm_hw_free(struct snd_pcm_substream *subs) { - return snd_pcm_free_vmalloc_buffer(subs); + return snd_pcm_lib_free_vmalloc_buffer(subs); } /* @@ -319,7 +276,7 @@ static struct snd_pcm_ops pdacf_pcm_capture_ops = { .prepare = pdacf_pcm_prepare, .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v0.10.2 From 6cedf8696d6a01bba391bdae06231243cfe2f48a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:30:24 +0100 Subject: sound: sgio2audio: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index f1d9d16..9b486be 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include #include @@ -603,25 +602,14 @@ static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_pcm_runtime *runtime = substream->runtime; - int size = params_buffer_bytes(hw_params); - - /* alloc virtual 'dma' area */ - if (runtime->dma_area) - vfree(runtime->dma_area); - runtime->dma_area = vmalloc_user(size); - if (runtime->dma_area == NULL) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } /* hw_free callback */ static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) { - vfree(substream->runtime->dma_area); - substream->runtime->dma_area = NULL; - return 0; + return snd_pcm_lib_free_vmalloc_buffer(substream); } /* prepare callback */ @@ -692,13 +680,6 @@ snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) chip->channel[chan->idx].pos); } -/* get the physical page pointer on the given offset */ -static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream, - unsigned long offset) -{ - return vmalloc_to_page(substream->runtime->dma_area + offset); -} - /* operators */ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .open = snd_sgio2audio_playback1_open, @@ -709,7 +690,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -721,7 +702,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -733,7 +714,7 @@ static struct snd_pcm_ops snd_sgio2audio_capture_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; /* -- cgit v0.10.2 From 149feef54bf543448dd4ec5820ef8ae178021c3a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:30:55 +0100 Subject: sound: vx: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 6644d00..c8385d2 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -46,7 +46,6 @@ */ #include -#include #include #include #include @@ -56,55 +55,6 @@ /* - * we use a vmalloc'ed (sg-)buffer - */ - -/* get the physical page pointer on the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* - * allocate a buffer via vmalloc_32(). - * called from hw_params - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - /* already allocated */ - if (runtime->dma_bytes >= size) - return 0; /* already enough large */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_32(size); - if (! runtime->dma_area) - return -ENOMEM; - memset(runtime->dma_area, 0, size); - runtime->dma_bytes = size; - return 1; /* changed */ -} - -/* - * free the buffer. - * called from hw_free callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - - -/* * read three pending pcm bytes via inb() */ static void vx_pcm_read_per_bytes(struct vx_core *chip, struct snd_pcm_runtime *runtime, @@ -865,7 +815,8 @@ static snd_pcm_uframes_t vx_pcm_playback_pointer(struct snd_pcm_substream *subs) static int vx_pcm_hw_params(struct snd_pcm_substream *subs, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_32_buffer + (subs, params_buffer_bytes(hw_params)); } /* @@ -873,7 +824,7 @@ static int vx_pcm_hw_params(struct snd_pcm_substream *subs, */ static int vx_pcm_hw_free(struct snd_pcm_substream *subs) { - return snd_pcm_free_vmalloc_buffer(subs); + return snd_pcm_lib_free_vmalloc_buffer(subs); } /* @@ -953,7 +904,7 @@ static struct snd_pcm_ops vx_pcm_playback_ops = { .prepare = vx_pcm_prepare, .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; @@ -1173,7 +1124,7 @@ static struct snd_pcm_ops vx_pcm_capture_ops = { .prepare = vx_pcm_prepare, .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v0.10.2 From c55675e348d9630c1ca69a190529bed1108c649d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:31:31 +0100 Subject: sound: usb-audio: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index af8869a..31b63ea 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -44,7 +44,6 @@ #include #include #include -#include #include #include #include @@ -735,41 +734,6 @@ static void snd_complete_sync_urb(struct urb *urb) } -/* get the physical page pointer at the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* allocate virtual buffer; may be called more than once */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already large enough */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_user(size); - if (!runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* free virtual buffer; may be called more than once */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - - /* * unlink active urbs. */ @@ -1449,8 +1413,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, unsigned int channels, rate, format; int ret, changed; - ret = snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); if (ret < 0) return ret; @@ -1507,7 +1471,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->period_bytes = 0; if (!subs->stream->chip->shutdown) release_substream_urbs(subs, 0); - return snd_pcm_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_vmalloc_buffer(substream); } /* @@ -1973,7 +1937,7 @@ static struct snd_pcm_ops snd_usb_playback_ops = { .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_pcm_playback_trigger, .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_usb_capture_ops = { @@ -1985,7 +1949,7 @@ static struct snd_pcm_ops snd_usb_capture_ops = { .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_pcm_capture_trigger, .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v0.10.2 From 5b4b2a41a1a80f5560364b7ef001486cd8fb5230 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:32:00 +0100 Subject: sound: ua101: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index ab9f8a2..16dc7bd 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include @@ -145,42 +144,6 @@ static struct usb_driver ua101_driver; static void abort_alsa_playback(struct ua101 *ua); static void abort_alsa_capture(struct ua101 *ua); -/* allocate virtual buffer; may be called more than once */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, - size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already large enough */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_user(size); - if (!runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* free virtual buffer; may be called more than once */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - -/* get the physical page pointer at the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - static const char *usb_error_string(int err) { switch (err) { @@ -791,8 +754,8 @@ static int capture_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; - return snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } static int playback_pcm_hw_params(struct snd_pcm_substream *substream, @@ -809,14 +772,13 @@ static int playback_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; - return snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } static int ua101_pcm_hw_free(struct snd_pcm_substream *substream) { - snd_pcm_free_vmalloc_buffer(substream); - return 0; + return snd_pcm_lib_free_vmalloc_buffer(substream); } static int capture_pcm_prepare(struct snd_pcm_substream *substream) @@ -948,7 +910,7 @@ static struct snd_pcm_ops capture_pcm_ops = { .prepare = capture_pcm_prepare, .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops playback_pcm_ops = { @@ -960,7 +922,7 @@ static struct snd_pcm_ops playback_pcm_ops = { .prepare = playback_pcm_prepare, .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static const struct uac_format_type_i_discrete_descriptor * -- cgit v0.10.2 From b35a28af0a64a1e8e389bc009b76253256d8fe7b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Dec 2009 12:00:22 +0000 Subject: ASoC: Add initial WM8955 CODEC driver The WM8955 is a low power, high quality stereo DAC with integrated headphone and loudspeaker amplifiers, designed to reduce external component requirements in portable digital audio applications. This is an initial driver implementing support for the majority of the functionality in the device, currently OUT3 is not supported. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/wm8955.h b/include/sound/wm8955.h new file mode 100644 index 0000000..5074ef4 --- /dev/null +++ b/include/sound/wm8955.h @@ -0,0 +1,26 @@ +/* + * Platform data for WM8955 + * + * Copyright 2009 Wolfson Microelectronics PLC. + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __WM8955_PDATA_H__ +#define __WM8955_PDATA_H__ + +struct wm8955_pdata { + /* Configure LOUT2/ROUT2 to drive a speaker */ + unsigned int out2_speaker:1; + + /* Configure MONOIN+/- in differential mode */ + unsigned int monoin_diff:1; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 691abe7..62ff26a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -52,6 +52,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C + select SND_SOC_WM8955 if I2C select SND_SOC_WM8960 if I2C select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C @@ -214,6 +215,9 @@ config SND_SOC_WM8904 config SND_SOC_WM8940 tristate +config SND_SOC_WM8955 + tristate + config SND_SOC_WM8960 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c0fd3c8..ea98354 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -39,6 +39,7 @@ snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o snd-soc-wm8940-objs := wm8940.o +snd-soc-wm8955-objs := wm8955.o snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o @@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o +obj-$(CONFIG_SND_SOC_WM8955) += snd-soc-wm8955.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c new file mode 100644 index 0000000..615dab2 --- /dev/null +++ b/sound/soc/codecs/wm8955.c @@ -0,0 +1,1151 @@ +/* + * wm8955.c -- WM8955 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8955.h" + +static struct snd_soc_codec *wm8955_codec; +struct snd_soc_codec_device soc_codec_dev_wm8955; + +#define WM8955_NUM_SUPPLIES 4 +static const char *wm8955_supply_names[WM8955_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD", + "HPVDD", + "AVDD", +}; + +/* codec private data */ +struct wm8955_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8955_MAX_REGISTER + 1]; + + unsigned int mclk_rate; + + int deemph; + int fs; + + struct regulator_bulk_data supplies[WM8955_NUM_SUPPLIES]; + + struct wm8955_pdata *pdata; +}; + +static const u16 wm8955_reg[WM8955_MAX_REGISTER + 1] = { + 0x0000, /* R0 */ + 0x0000, /* R1 */ + 0x0079, /* R2 - LOUT1 volume */ + 0x0079, /* R3 - ROUT1 volume */ + 0x0000, /* R4 */ + 0x0008, /* R5 - DAC Control */ + 0x0000, /* R6 */ + 0x000A, /* R7 - Audio Interface */ + 0x0000, /* R8 - Sample Rate */ + 0x0000, /* R9 */ + 0x00FF, /* R10 - Left DAC volume */ + 0x00FF, /* R11 - Right DAC volume */ + 0x000F, /* R12 - Bass control */ + 0x000F, /* R13 - Treble control */ + 0x0000, /* R14 */ + 0x0000, /* R15 - Reset */ + 0x0000, /* R16 */ + 0x0000, /* R17 */ + 0x0000, /* R18 */ + 0x0000, /* R19 */ + 0x0000, /* R20 */ + 0x0000, /* R21 */ + 0x0000, /* R22 */ + 0x00C1, /* R23 - Additional control (1) */ + 0x0000, /* R24 - Additional control (2) */ + 0x0000, /* R25 - Power Management (1) */ + 0x0000, /* R26 - Power Management (2) */ + 0x0000, /* R27 - Additional Control (3) */ + 0x0000, /* R28 */ + 0x0000, /* R29 */ + 0x0000, /* R30 */ + 0x0000, /* R31 */ + 0x0000, /* R32 */ + 0x0000, /* R33 */ + 0x0050, /* R34 - Left out Mix (1) */ + 0x0050, /* R35 - Left out Mix (2) */ + 0x0050, /* R36 - Right out Mix (1) */ + 0x0050, /* R37 - Right Out Mix (2) */ + 0x0050, /* R38 - Mono out Mix (1) */ + 0x0050, /* R39 - Mono out Mix (2) */ + 0x0079, /* R40 - LOUT2 volume */ + 0x0079, /* R41 - ROUT2 volume */ + 0x0079, /* R42 - MONOOUT volume */ + 0x0000, /* R43 - Clocking / PLL */ + 0x0103, /* R44 - PLL Control 1 */ + 0x0024, /* R45 - PLL Control 2 */ + 0x01BA, /* R46 - PLL Control 3 */ + 0x0000, /* R47 */ + 0x0000, /* R48 */ + 0x0000, /* R49 */ + 0x0000, /* R50 */ + 0x0000, /* R51 */ + 0x0000, /* R52 */ + 0x0000, /* R53 */ + 0x0000, /* R54 */ + 0x0000, /* R55 */ + 0x0000, /* R56 */ + 0x0000, /* R57 */ + 0x0000, /* R58 */ + 0x0000, /* R59 - PLL Control 4 */ +}; + +static int wm8955_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8955_RESET, 0); +} + +struct pll_factors { + int n; + int k; + int outdiv; +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 22) * 10) + +static int wm8995_pll_factors(struct device *dev, + int Fref, int Fout, struct pll_factors *pll) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + + dev_dbg(dev, "Fref=%u Fout=%u\n", Fref, Fout); + + /* The oscilator should run at should be 90-100MHz, and + * there's a divide by 4 plus an optional divide by 2 in the + * output path to generate the system clock. The clock table + * is sortd so we should always generate a suitable target. */ + target = Fout * 4; + if (target < 90000000) { + pll->outdiv = 1; + target *= 2; + } else { + pll->outdiv = 0; + } + + WARN_ON(target < 90000000 || target > 100000000); + + dev_dbg(dev, "Fvco=%dHz\n", target); + + /* Now, calculate N.K */ + Ndiv = target / Fref; + + pll->n = Ndiv; + Nmod = target % Fref; + dev_dbg(dev, "Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, Fref); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + pll->k = K / 10; + + dev_dbg(dev, "N=%x K=%x OUTDIV=%x\n", pll->n, pll->k, pll->outdiv); + + return 0; +} + +/* Lookup table specifiying SRATE (table 25 in datasheet); some of the + * output frequencies have been rounded to the standard frequencies + * they are intended to match where the error is slight. */ +static struct { + int mclk; + int fs; + int usb; + int sr; +} clock_cfgs[] = { + { 18432000, 8000, 0, 3, }, + { 18432000, 12000, 0, 9, }, + { 18432000, 16000, 0, 11, }, + { 18432000, 24000, 0, 29, }, + { 18432000, 32000, 0, 13, }, + { 18432000, 48000, 0, 1, }, + { 18432000, 96000, 0, 15, }, + + { 16934400, 8018, 0, 19, }, + { 16934400, 11025, 0, 25, }, + { 16934400, 22050, 0, 27, }, + { 16934400, 44100, 0, 17, }, + { 16934400, 88200, 0, 31, }, + + { 12000000, 8000, 1, 2, }, + { 12000000, 11025, 1, 25, }, + { 12000000, 12000, 1, 8, }, + { 12000000, 16000, 1, 10, }, + { 12000000, 22050, 1, 27, }, + { 12000000, 24000, 1, 28, }, + { 12000000, 32000, 1, 12, }, + { 12000000, 44100, 1, 17, }, + { 12000000, 48000, 1, 0, }, + { 12000000, 88200, 1, 31, }, + { 12000000, 96000, 1, 14, }, + + { 12288000, 8000, 0, 2, }, + { 12288000, 12000, 0, 8, }, + { 12288000, 16000, 0, 10, }, + { 12288000, 24000, 0, 28, }, + { 12288000, 32000, 0, 12, }, + { 12288000, 48000, 0, 0, }, + { 12288000, 96000, 0, 14, }, + + { 12289600, 8018, 0, 18, }, + { 12289600, 11025, 0, 24, }, + { 12289600, 22050, 0, 26, }, + { 11289600, 44100, 0, 16, }, + { 11289600, 88200, 0, 31, }, +}; + +static int wm8955_configure_clocking(struct snd_soc_codec *codec) +{ + struct wm8955_priv *wm8955 = codec->private_data; + int i, ret, val; + int clocking = 0; + int srate = 0; + int sr = -1; + struct pll_factors pll; + + /* If we're not running a sample rate currently just pick one */ + if (wm8955->fs == 0) + wm8955->fs = 8000; + + /* Can we generate an exact output? */ + for (i = 0; i < ARRAY_SIZE(clock_cfgs); i++) { + if (wm8955->fs != clock_cfgs[i].fs) + continue; + sr = i; + + if (wm8955->mclk_rate == clock_cfgs[i].mclk) + break; + } + + /* We should never get here with an unsupported sample rate */ + if (sr == -1) { + dev_err(codec->dev, "Sample rate %dHz unsupported\n", + wm8955->fs); + WARN_ON(sr == -1); + return -EINVAL; + } + + if (i == ARRAY_SIZE(clock_cfgs)) { + /* If we can't generate the right clock from MCLK then + * we should configure the PLL to supply us with an + * appropriate clock. + */ + clocking |= WM8955_MCLKSEL; + + /* Use the last divider configuration we saw for the + * sample rate. */ + ret = wm8995_pll_factors(codec->dev, wm8955->mclk_rate, + clock_cfgs[sr].mclk, &pll); + if (ret != 0) { + dev_err(codec->dev, + "Unable to generate %dHz from %dHz MCLK\n", + wm8955->fs, wm8955->mclk_rate); + return -EINVAL; + } + + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_1, + WM8955_N_MASK | WM8955_K_21_18_MASK, + (pll.n << WM8955_N_SHIFT) | + pll.k >> 18); + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_2, + WM8955_K_17_9_MASK, + (pll.k >> 9) & WM8955_K_17_9_MASK); + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_2, + WM8955_K_8_0_MASK, + pll.k & WM8955_K_8_0_MASK); + if (pll.k) + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_4, + WM8955_KEN, WM8955_KEN); + else + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_4, + WM8955_KEN, 0); + + if (pll.outdiv) + val = WM8955_PLL_RB | WM8955_PLLOUTDIV2; + else + val = WM8955_PLL_RB; + + /* Now start the PLL running */ + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_PLL_RB | WM8955_PLLOUTDIV2, val); + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_PLLEN, WM8955_PLLEN); + } + + srate = clock_cfgs[sr].usb | (clock_cfgs[sr].sr << WM8955_SR_SHIFT); + + snd_soc_update_bits(codec, WM8955_SAMPLE_RATE, + WM8955_USB | WM8955_SR_MASK, srate); + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_MCLKSEL, clocking); + + return 0; +} + +static int wm8955_sysclk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + int ret = 0; + + /* Always disable the clocks - if we're doing reconfiguration this + * avoids misclocking. + */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_DIGENB, 0); + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_PLL_RB | WM8955_PLLEN, 0); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + break; + case SND_SOC_DAPM_PRE_PMU: + ret = wm8955_configure_clocking(codec); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int wm8955_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8955_priv *wm8955 = codec->private_data; + int val, i, best; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8955->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - wm8955->fs) < + abs(deemph_settings[best] - wm8955->fs)) + best = i; + } + + val = best << WM8955_DEEMPH_SHIFT; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, WM8955_DAC_CONTROL, + WM8955_DEEMPH_MASK, val); +} + +static int wm8955_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8955_priv *wm8955 = codec->private_data; + + return wm8955->deemph; +} + +static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8955_priv *wm8955 = codec->private_data; + int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + wm8955->deemph = deemph; + + return wm8955_set_deemph(codec); +} + +static const char *bass_mode_text[] = { + "Linear", "Adaptive", +}; + +static const struct soc_enum bass_mode = + SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 7, 2, bass_mode_text); + +static const char *bass_cutoff_text[] = { + "Low", "High" +}; + +static const struct soc_enum bass_cutoff = + SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 6, 2, bass_cutoff_text); + +static const char *treble_cutoff_text[] = { + "High", "Low" +}; + +static const struct soc_enum treble_cutoff = + SOC_ENUM_SINGLE(WM8955_TREBLE_CONTROL, 6, 2, treble_cutoff_text); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(atten_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -2100, 300, 0); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(treble_tlv, -1200, 150, 1); + +static const struct snd_kcontrol_new wm8955_snd_controls[] = { +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8955_LEFT_DAC_VOLUME, + WM8955_RIGHT_DAC_VOLUME, 0, 255, 0, digital_tlv), +SOC_SINGLE_TLV("Playback Attenuation Volume", WM8955_DAC_CONTROL, 7, 1, 1, + atten_tlv), +SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + wm8955_get_deemph, wm8955_put_deemph), + +SOC_ENUM("Bass Mode", bass_mode), +SOC_ENUM("Bass Cutoff", bass_cutoff), +SOC_SINGLE("Bass Volume", WM8955_BASS_CONTROL, 0, 15, 1), + +SOC_ENUM("Treble Cutoff", treble_cutoff), +SOC_SINGLE_TLV("Treble Volume", WM8955_TREBLE_CONTROL, 0, 14, 1, treble_tlv), + +SOC_SINGLE_TLV("Left Bypass Volume", WM8955_LEFT_OUT_MIX_1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Left Mono Volume", WM8955_LEFT_OUT_MIX_2, 4, 7, 1, + bypass_tlv), + +SOC_SINGLE_TLV("Right Mono Volume", WM8955_RIGHT_OUT_MIX_1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Right Bypass Volume", WM8955_RIGHT_OUT_MIX_2, 4, 7, 1, + bypass_tlv), + +/* Not a stereo pair so they line up with the DAPM switches */ +SOC_SINGLE_TLV("Mono Left Bypass Volume", WM8955_MONO_OUT_MIX_1, 4, 7, 1, + mono_tlv), +SOC_SINGLE_TLV("Mono Right Bypass Volume", WM8955_MONO_OUT_MIX_2, 4, 7, 1, + mono_tlv), + +SOC_DOUBLE_R_TLV("Headphone Volume", WM8955_LOUT1_VOLUME, + WM8955_ROUT1_VOLUME, 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Headphone ZC Switch", WM8955_LOUT1_VOLUME, + WM8955_ROUT1_VOLUME, 7, 1, 0), + +SOC_DOUBLE_R_TLV("Speaker Volume", WM8955_LOUT2_VOLUME, + WM8955_ROUT2_VOLUME, 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Speaker ZC Switch", WM8955_LOUT2_VOLUME, + WM8955_ROUT2_VOLUME, 7, 1, 0), + +SOC_SINGLE_TLV("Mono Volume", WM8955_MONOOUT_VOLUME, 0, 127, 0, out_tlv), +SOC_SINGLE("Mono ZC Switch", WM8955_MONOOUT_VOLUME, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lmixer[] = { +SOC_DAPM_SINGLE("Playback Switch", WM8955_LEFT_OUT_MIX_1, 8, 1, 0), +SOC_DAPM_SINGLE("Bypass Switch", WM8955_LEFT_OUT_MIX_1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Playback Switch", WM8955_LEFT_OUT_MIX_2, 8, 1, 0), +SOC_DAPM_SINGLE("Mono Switch", WM8955_LEFT_OUT_MIX_2, 7, 1, 0), +}; + +static const struct snd_kcontrol_new rmixer[] = { +SOC_DAPM_SINGLE("Left Playback Switch", WM8955_RIGHT_OUT_MIX_1, 8, 1, 0), +SOC_DAPM_SINGLE("Mono Switch", WM8955_RIGHT_OUT_MIX_1, 7, 1, 0), +SOC_DAPM_SINGLE("Playback Switch", WM8955_RIGHT_OUT_MIX_2, 8, 1, 0), +SOC_DAPM_SINGLE("Bypass Switch", WM8955_RIGHT_OUT_MIX_2, 7, 1, 0), +}; + +static const struct snd_kcontrol_new mmixer[] = { +SOC_DAPM_SINGLE("Left Playback Switch", WM8955_MONO_OUT_MIX_1, 8, 1, 0), +SOC_DAPM_SINGLE("Left Bypass Switch", WM8955_MONO_OUT_MIX_1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Playback Switch", WM8955_MONO_OUT_MIX_2, 8, 1, 0), +SOC_DAPM_SINGLE("Right Bypass Switch", WM8955_MONO_OUT_MIX_2, 7, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8955_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MONOIN-"), +SND_SOC_DAPM_INPUT("MONOIN+"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("LINEINL"), + +SND_SOC_DAPM_PGA("Mono Input", SND_SOC_NOPM, 0, 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("SYSCLK", WM8955_POWER_MANAGEMENT_1, 0, 1, wm8955_sysclk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("TSDEN", WM8955_ADDITIONAL_CONTROL_1, 8, 0, NULL, 0), + +SND_SOC_DAPM_DAC("DACL", "Playback", WM8955_POWER_MANAGEMENT_2, 8, 0), +SND_SOC_DAPM_DAC("DACR", "Playback", WM8955_POWER_MANAGEMENT_2, 7, 0), + +SND_SOC_DAPM_PGA("LOUT1 PGA", WM8955_POWER_MANAGEMENT_2, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("ROUT1 PGA", WM8955_POWER_MANAGEMENT_2, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("LOUT2 PGA", WM8955_POWER_MANAGEMENT_2, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("ROUT2 PGA", WM8955_POWER_MANAGEMENT_2, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MOUT PGA", WM8955_POWER_MANAGEMENT_2, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("OUT3 PGA", WM8955_POWER_MANAGEMENT_2, 1, 0, NULL, 0), + +/* The names are chosen to make the control names nice */ +SND_SOC_DAPM_MIXER("Left", SND_SOC_NOPM, 0, 0, + lmixer, ARRAY_SIZE(lmixer)), +SND_SOC_DAPM_MIXER("Right", SND_SOC_NOPM, 0, 0, + rmixer, ARRAY_SIZE(rmixer)), +SND_SOC_DAPM_MIXER("Mono", SND_SOC_NOPM, 0, 0, + mmixer, ARRAY_SIZE(mmixer)), + +SND_SOC_DAPM_OUTPUT("LOUT1"), +SND_SOC_DAPM_OUTPUT("ROUT1"), +SND_SOC_DAPM_OUTPUT("LOUT2"), +SND_SOC_DAPM_OUTPUT("ROUT2"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("OUT3"), +}; + +static const struct snd_soc_dapm_route wm8955_intercon[] = { + { "DACL", NULL, "SYSCLK" }, + { "DACR", NULL, "SYSCLK" }, + + { "Mono Input", NULL, "MONOIN-" }, + { "Mono Input", NULL, "MONOIN+" }, + + { "Left", "Playback Switch", "DACL" }, + { "Left", "Right Playback Switch", "DACR" }, + { "Left", "Bypass Switch", "LINEINL" }, + { "Left", "Mono Switch", "Mono Input" }, + + { "Right", "Playback Switch", "DACR" }, + { "Right", "Left Playback Switch", "DACL" }, + { "Right", "Bypass Switch", "LINEINR" }, + { "Right", "Mono Switch", "Mono Input" }, + + { "Mono", "Left Playback Switch", "DACL" }, + { "Mono", "Right Playback Switch", "DACR" }, + { "Mono", "Left Bypass Switch", "LINEINL" }, + { "Mono", "Right Bypass Switch", "LINEINR" }, + + { "LOUT1 PGA", NULL, "Left" }, + { "LOUT1", NULL, "TSDEN" }, + { "LOUT1", NULL, "LOUT1 PGA" }, + + { "ROUT1 PGA", NULL, "Right" }, + { "ROUT1", NULL, "TSDEN" }, + { "ROUT1", NULL, "ROUT1 PGA" }, + + { "LOUT2 PGA", NULL, "Left" }, + { "LOUT2", NULL, "TSDEN" }, + { "LOUT2", NULL, "LOUT2 PGA" }, + + { "ROUT2 PGA", NULL, "Right" }, + { "ROUT2", NULL, "TSDEN" }, + { "ROUT2", NULL, "ROUT2 PGA" }, + + { "MOUT PGA", NULL, "Mono" }, + { "MONOOUT", NULL, "MOUT PGA" }, + + /* OUT3 not currently implemented */ + { "OUT3", NULL, "OUT3 PGA" }, +}; + +static int wm8955_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_add_controls(codec, wm8955_snd_controls, + ARRAY_SIZE(wm8955_snd_controls)); + + snd_soc_dapm_new_controls(codec, wm8955_dapm_widgets, + ARRAY_SIZE(wm8955_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, wm8955_intercon, + ARRAY_SIZE(wm8955_intercon)); + + return 0; +} + +static int wm8955_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8955_priv *wm8955 = codec->private_data; + int ret; + int wl; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wl = 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wl = 0x4; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wl = 0x8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wl = 0xc; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, WM8955_AUDIO_INTERFACE, + WM8955_WL_MASK, wl); + + wm8955->fs = params_rate(params); + wm8955_set_deemph(codec); + + /* If the chip is clocked then disable the clocks and force a + * reconfiguration, otherwise DAPM will power up the + * clocks for us later. */ + ret = snd_soc_read(codec, WM8955_POWER_MANAGEMENT_1); + if (ret < 0) + return ret; + if (ret & WM8955_DIGENB) { + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_DIGENB, 0); + snd_soc_update_bits(codec, WM8955_CLOCKING_PLL, + WM8955_PLL_RB | WM8955_PLLEN, 0); + + wm8955_configure_clocking(codec); + } + + return 0; +} + + +static int wm8955_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8955_priv *priv = codec->private_data; + int div; + + switch (clk_id) { + case WM8955_CLK_MCLK: + if (freq > 15000000) { + priv->mclk_rate = freq /= 2; + div = WM8955_MCLKDIV2; + } else { + priv->mclk_rate = freq; + div = 0; + } + + snd_soc_update_bits(codec, WM8955_SAMPLE_RATE, + WM8955_MCLKDIV2, div); + break; + + default: + return -EINVAL; + } + + dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq); + + return 0; +} + +static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u16 aif = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif |= WM8955_MS; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif |= WM8955_LRP; + case SND_SOC_DAIFMT_DSP_A: + aif |= 0x3; + break; + case SND_SOC_DAIFMT_I2S: + aif |= 0x2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif |= 0x1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif |= WM8955_BCLKINV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif |= WM8955_BCLKINV | WM8955_LRP; + break; + case SND_SOC_DAIFMT_IB_NF: + aif |= WM8955_BCLKINV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif |= WM8955_LRP; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, WM8955_AUDIO_INTERFACE, + WM8955_MS | WM8955_FORMAT_MASK | WM8955_BCLKINV | + WM8955_LRP, aif); + + return 0; +} + + +static int wm8955_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int val; + + if (mute) + val = WM8955_DACMU; + else + val = 0; + + snd_soc_update_bits(codec, WM8955_DAC_CONTROL, WM8955_DACMU, val); + + return 0; +} + +static int wm8955_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8955_priv *wm8955 = codec->private_data; + int ret, i; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID resistance 2*50k */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_VMIDSEL_MASK, + 0x1 << WM8955_VMIDSEL_SHIFT); + + /* Default bias current */ + snd_soc_update_bits(codec, WM8955_ADDITIONAL_CONTROL_1, + WM8955_VSEL_MASK, + 0x2 << WM8955_VSEL_SHIFT); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8955->supplies), + wm8955->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + /* Sync back cached values if they're + * different from the hardware default. + */ + for (i = 0; i < ARRAY_SIZE(wm8955->reg_cache); i++) { + if (i == WM8955_RESET) + continue; + + if (wm8955->reg_cache[i] == wm8955_reg[i]) + continue; + + snd_soc_write(codec, i, wm8955->reg_cache[i]); + } + + /* Enable VREF and VMID */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_VREF | + WM8955_VMIDSEL_MASK, + WM8955_VREF | + 0x3 << WM8955_VREF_SHIFT); + + /* Let VMID ramp */ + msleep(500); + + /* High resistance VROI to maintain outputs */ + snd_soc_update_bits(codec, + WM8955_ADDITIONAL_CONTROL_3, + WM8955_VROI, WM8955_VROI); + } + + /* Maintain VMID with 2*250k */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_VMIDSEL_MASK, + 0x2 << WM8955_VMIDSEL_SHIFT); + + /* Minimum bias current */ + snd_soc_update_bits(codec, WM8955_ADDITIONAL_CONTROL_1, + WM8955_VSEL_MASK, 0); + break; + + case SND_SOC_BIAS_OFF: + /* Low resistance VROI to help discharge */ + snd_soc_update_bits(codec, + WM8955_ADDITIONAL_CONTROL_3, + WM8955_VROI, 0); + + /* Turn off VMID and VREF */ + snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, + WM8955_VREF | + WM8955_VMIDSEL_MASK, 0); + + regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), + wm8955->supplies); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8955_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8955_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8955_dai_ops = { + .set_sysclk = wm8955_set_sysclk, + .set_fmt = wm8955_set_fmt, + .hw_params = wm8955_hw_params, + .digital_mute = wm8955_digital_mute, +}; + +struct snd_soc_dai wm8955_dai = { + .name = "WM8955", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8955_RATES, + .formats = WM8955_FORMATS, + }, + .ops = &wm8955_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8955_dai); + +#ifdef CONFIG_PM +static int wm8955_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8955_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8955_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8955_suspend NULL +#define wm8955_resume NULL +#endif + +static int wm8955_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8955_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8955_codec; + codec = wm8955_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + wm8955_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +static int wm8955_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8955 = { + .probe = wm8955_probe, + .remove = wm8955_remove, + .suspend = wm8955_suspend, + .resume = wm8955_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8955); + +static int wm8955_register(struct wm8955_priv *wm8955, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8955->codec; + int i; + + if (wm8955_codec) { + dev_err(codec->dev, "Another WM8955 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8955; + codec->name = "WM8955"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8955_set_bias_level; + codec->dai = &wm8955_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8955_MAX_REGISTER; + codec->reg_cache = &wm8955->reg_cache; + + memcpy(codec->reg_cache, wm8955_reg, sizeof(wm8955_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8955->supplies); i++) + wm8955->supplies[i].supply = wm8955_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8955->supplies), + wm8955->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8955->supplies), + wm8955->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = wm8955_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + goto err_enable; + } + + wm8955_dai.dev = codec->dev; + + /* Change some default settings - latch VU and enable ZC */ + wm8955->reg_cache[WM8955_LEFT_DAC_VOLUME] |= WM8955_LDVU; + wm8955->reg_cache[WM8955_RIGHT_DAC_VOLUME] |= WM8955_RDVU; + wm8955->reg_cache[WM8955_LOUT1_VOLUME] |= WM8955_LO1VU | WM8955_LO1ZC; + wm8955->reg_cache[WM8955_ROUT1_VOLUME] |= WM8955_RO1VU | WM8955_RO1ZC; + wm8955->reg_cache[WM8955_LOUT2_VOLUME] |= WM8955_LO2VU | WM8955_LO2ZC; + wm8955->reg_cache[WM8955_ROUT2_VOLUME] |= WM8955_RO2VU | WM8955_RO2ZC; + wm8955->reg_cache[WM8955_MONOOUT_VOLUME] |= WM8955_MOZC; + + /* Also enable adaptive bass boost by default */ + wm8955->reg_cache[WM8955_BASS_CONTROL] |= WM8955_BB; + + /* Set platform data values */ + if (wm8955->pdata) { + if (wm8955->pdata->out2_speaker) + wm8955->reg_cache[WM8955_ADDITIONAL_CONTROL_2] + |= WM8955_ROUT2INV; + + if (wm8955->pdata->monoin_diff) + wm8955->reg_cache[WM8955_MONO_OUT_MIX_1] + |= WM8955_DMEN; + } + + wm8955_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); + + wm8955_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8955_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); +err: + kfree(wm8955); + return ret; +} + +static void wm8955_unregister(struct wm8955_priv *wm8955) +{ + wm8955_set_bias_level(&wm8955->codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); + snd_soc_unregister_dai(&wm8955_dai); + snd_soc_unregister_codec(&wm8955->codec); + kfree(wm8955); + wm8955_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8955_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8955_priv *wm8955; + struct snd_soc_codec *codec; + + wm8955 = kzalloc(sizeof(struct wm8955_priv), GFP_KERNEL); + if (wm8955 == NULL) + return -ENOMEM; + + codec = &wm8955->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8955); + codec->control_data = i2c; + wm8955->pdata = i2c->dev.platform_data; + + codec->dev = &i2c->dev; + + return wm8955_register(wm8955, SND_SOC_I2C); +} + +static __devexit int wm8955_i2c_remove(struct i2c_client *client) +{ + struct wm8955_priv *wm8955 = i2c_get_clientdata(client); + wm8955_unregister(wm8955); + return 0; +} + +static const struct i2c_device_id wm8955_i2c_id[] = { + { "wm8955", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); + +static struct i2c_driver wm8955_i2c_driver = { + .driver = { + .name = "wm8955", + .owner = THIS_MODULE, + }, + .probe = wm8955_i2c_probe, + .remove = __devexit_p(wm8955_i2c_remove), + .id_table = wm8955_i2c_id, +}; +#endif + +static int __init wm8955_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8955_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8955 I2C driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8955_modinit); + +static void __exit wm8955_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8955_i2c_driver); +#endif +} +module_exit(wm8955_exit); + +MODULE_DESCRIPTION("ASoC WM8955 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8955.h b/sound/soc/codecs/wm8955.h new file mode 100644 index 0000000..ae349c8 --- /dev/null +++ b/sound/soc/codecs/wm8955.h @@ -0,0 +1,489 @@ +/* + * wm8955.h -- WM8904 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics, plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8955_H +#define _WM8955_H + +#define WM8955_CLK_MCLK 1 + +extern struct snd_soc_dai wm8955_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8955; + +/* + * Register values. + */ +#define WM8955_LOUT1_VOLUME 0x02 +#define WM8955_ROUT1_VOLUME 0x03 +#define WM8955_DAC_CONTROL 0x05 +#define WM8955_AUDIO_INTERFACE 0x07 +#define WM8955_SAMPLE_RATE 0x08 +#define WM8955_LEFT_DAC_VOLUME 0x0A +#define WM8955_RIGHT_DAC_VOLUME 0x0B +#define WM8955_BASS_CONTROL 0x0C +#define WM8955_TREBLE_CONTROL 0x0D +#define WM8955_RESET 0x0F +#define WM8955_ADDITIONAL_CONTROL_1 0x17 +#define WM8955_ADDITIONAL_CONTROL_2 0x18 +#define WM8955_POWER_MANAGEMENT_1 0x19 +#define WM8955_POWER_MANAGEMENT_2 0x1A +#define WM8955_ADDITIONAL_CONTROL_3 0x1B +#define WM8955_LEFT_OUT_MIX_1 0x22 +#define WM8955_LEFT_OUT_MIX_2 0x23 +#define WM8955_RIGHT_OUT_MIX_1 0x24 +#define WM8955_RIGHT_OUT_MIX_2 0x25 +#define WM8955_MONO_OUT_MIX_1 0x26 +#define WM8955_MONO_OUT_MIX_2 0x27 +#define WM8955_LOUT2_VOLUME 0x28 +#define WM8955_ROUT2_VOLUME 0x29 +#define WM8955_MONOOUT_VOLUME 0x2A +#define WM8955_CLOCKING_PLL 0x2B +#define WM8955_PLL_CONTROL_1 0x2C +#define WM8955_PLL_CONTROL_2 0x2D +#define WM8955_PLL_CONTROL_3 0x2E +#define WM8955_PLL_CONTROL_4 0x3B + +#define WM8955_REGISTER_COUNT 29 +#define WM8955_MAX_REGISTER 0x3B + +/* + * Field Definitions. + */ + +/* + * R2 (0x02) - LOUT1 volume + */ +#define WM8955_LO1VU 0x0100 /* LO1VU */ +#define WM8955_LO1VU_MASK 0x0100 /* LO1VU */ +#define WM8955_LO1VU_SHIFT 8 /* LO1VU */ +#define WM8955_LO1VU_WIDTH 1 /* LO1VU */ +#define WM8955_LO1ZC 0x0080 /* LO1ZC */ +#define WM8955_LO1ZC_MASK 0x0080 /* LO1ZC */ +#define WM8955_LO1ZC_SHIFT 7 /* LO1ZC */ +#define WM8955_LO1ZC_WIDTH 1 /* LO1ZC */ +#define WM8955_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */ +#define WM8955_LOUTVOL_SHIFT 0 /* LOUTVOL - [6:0] */ +#define WM8955_LOUTVOL_WIDTH 7 /* LOUTVOL - [6:0] */ + +/* + * R3 (0x03) - ROUT1 volume + */ +#define WM8955_RO1VU 0x0100 /* RO1VU */ +#define WM8955_RO1VU_MASK 0x0100 /* RO1VU */ +#define WM8955_RO1VU_SHIFT 8 /* RO1VU */ +#define WM8955_RO1VU_WIDTH 1 /* RO1VU */ +#define WM8955_RO1ZC 0x0080 /* RO1ZC */ +#define WM8955_RO1ZC_MASK 0x0080 /* RO1ZC */ +#define WM8955_RO1ZC_SHIFT 7 /* RO1ZC */ +#define WM8955_RO1ZC_WIDTH 1 /* RO1ZC */ +#define WM8955_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */ +#define WM8955_ROUTVOL_SHIFT 0 /* ROUTVOL - [6:0] */ +#define WM8955_ROUTVOL_WIDTH 7 /* ROUTVOL - [6:0] */ + +/* + * R5 (0x05) - DAC Control + */ +#define WM8955_DAT 0x0080 /* DAT */ +#define WM8955_DAT_MASK 0x0080 /* DAT */ +#define WM8955_DAT_SHIFT 7 /* DAT */ +#define WM8955_DAT_WIDTH 1 /* DAT */ +#define WM8955_DACMU 0x0008 /* DACMU */ +#define WM8955_DACMU_MASK 0x0008 /* DACMU */ +#define WM8955_DACMU_SHIFT 3 /* DACMU */ +#define WM8955_DACMU_WIDTH 1 /* DACMU */ +#define WM8955_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */ +#define WM8955_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */ +#define WM8955_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */ + +/* + * R7 (0x07) - Audio Interface + */ +#define WM8955_BCLKINV 0x0080 /* BCLKINV */ +#define WM8955_BCLKINV_MASK 0x0080 /* BCLKINV */ +#define WM8955_BCLKINV_SHIFT 7 /* BCLKINV */ +#define WM8955_BCLKINV_WIDTH 1 /* BCLKINV */ +#define WM8955_MS 0x0040 /* MS */ +#define WM8955_MS_MASK 0x0040 /* MS */ +#define WM8955_MS_SHIFT 6 /* MS */ +#define WM8955_MS_WIDTH 1 /* MS */ +#define WM8955_LRSWAP 0x0020 /* LRSWAP */ +#define WM8955_LRSWAP_MASK 0x0020 /* LRSWAP */ +#define WM8955_LRSWAP_SHIFT 5 /* LRSWAP */ +#define WM8955_LRSWAP_WIDTH 1 /* LRSWAP */ +#define WM8955_LRP 0x0010 /* LRP */ +#define WM8955_LRP_MASK 0x0010 /* LRP */ +#define WM8955_LRP_SHIFT 4 /* LRP */ +#define WM8955_LRP_WIDTH 1 /* LRP */ +#define WM8955_WL_MASK 0x000C /* WL - [3:2] */ +#define WM8955_WL_SHIFT 2 /* WL - [3:2] */ +#define WM8955_WL_WIDTH 2 /* WL - [3:2] */ +#define WM8955_FORMAT_MASK 0x0003 /* FORMAT - [1:0] */ +#define WM8955_FORMAT_SHIFT 0 /* FORMAT - [1:0] */ +#define WM8955_FORMAT_WIDTH 2 /* FORMAT - [1:0] */ + +/* + * R8 (0x08) - Sample Rate + */ +#define WM8955_BCLKDIV2 0x0080 /* BCLKDIV2 */ +#define WM8955_BCLKDIV2_MASK 0x0080 /* BCLKDIV2 */ +#define WM8955_BCLKDIV2_SHIFT 7 /* BCLKDIV2 */ +#define WM8955_BCLKDIV2_WIDTH 1 /* BCLKDIV2 */ +#define WM8955_MCLKDIV2 0x0040 /* MCLKDIV2 */ +#define WM8955_MCLKDIV2_MASK 0x0040 /* MCLKDIV2 */ +#define WM8955_MCLKDIV2_SHIFT 6 /* MCLKDIV2 */ +#define WM8955_MCLKDIV2_WIDTH 1 /* MCLKDIV2 */ +#define WM8955_SR_MASK 0x003E /* SR - [5:1] */ +#define WM8955_SR_SHIFT 1 /* SR - [5:1] */ +#define WM8955_SR_WIDTH 5 /* SR - [5:1] */ +#define WM8955_USB 0x0001 /* USB */ +#define WM8955_USB_MASK 0x0001 /* USB */ +#define WM8955_USB_SHIFT 0 /* USB */ +#define WM8955_USB_WIDTH 1 /* USB */ + +/* + * R10 (0x0A) - Left DAC volume + */ +#define WM8955_LDVU 0x0100 /* LDVU */ +#define WM8955_LDVU_MASK 0x0100 /* LDVU */ +#define WM8955_LDVU_SHIFT 8 /* LDVU */ +#define WM8955_LDVU_WIDTH 1 /* LDVU */ +#define WM8955_LDACVOL_MASK 0x00FF /* LDACVOL - [7:0] */ +#define WM8955_LDACVOL_SHIFT 0 /* LDACVOL - [7:0] */ +#define WM8955_LDACVOL_WIDTH 8 /* LDACVOL - [7:0] */ + +/* + * R11 (0x0B) - Right DAC volume + */ +#define WM8955_RDVU 0x0100 /* RDVU */ +#define WM8955_RDVU_MASK 0x0100 /* RDVU */ +#define WM8955_RDVU_SHIFT 8 /* RDVU */ +#define WM8955_RDVU_WIDTH 1 /* RDVU */ +#define WM8955_RDACVOL_MASK 0x00FF /* RDACVOL - [7:0] */ +#define WM8955_RDACVOL_SHIFT 0 /* RDACVOL - [7:0] */ +#define WM8955_RDACVOL_WIDTH 8 /* RDACVOL - [7:0] */ + +/* + * R12 (0x0C) - Bass control + */ +#define WM8955_BB 0x0080 /* BB */ +#define WM8955_BB_MASK 0x0080 /* BB */ +#define WM8955_BB_SHIFT 7 /* BB */ +#define WM8955_BB_WIDTH 1 /* BB */ +#define WM8955_BC 0x0040 /* BC */ +#define WM8955_BC_MASK 0x0040 /* BC */ +#define WM8955_BC_SHIFT 6 /* BC */ +#define WM8955_BC_WIDTH 1 /* BC */ +#define WM8955_BASS_MASK 0x000F /* BASS - [3:0] */ +#define WM8955_BASS_SHIFT 0 /* BASS - [3:0] */ +#define WM8955_BASS_WIDTH 4 /* BASS - [3:0] */ + +/* + * R13 (0x0D) - Treble control + */ +#define WM8955_TC 0x0040 /* TC */ +#define WM8955_TC_MASK 0x0040 /* TC */ +#define WM8955_TC_SHIFT 6 /* TC */ +#define WM8955_TC_WIDTH 1 /* TC */ +#define WM8955_TRBL_MASK 0x000F /* TRBL - [3:0] */ +#define WM8955_TRBL_SHIFT 0 /* TRBL - [3:0] */ +#define WM8955_TRBL_WIDTH 4 /* TRBL - [3:0] */ + +/* + * R15 (0x0F) - Reset + */ +#define WM8955_RESET_MASK 0x01FF /* RESET - [8:0] */ +#define WM8955_RESET_SHIFT 0 /* RESET - [8:0] */ +#define WM8955_RESET_WIDTH 9 /* RESET - [8:0] */ + +/* + * R23 (0x17) - Additional control (1) + */ +#define WM8955_TSDEN 0x0100 /* TSDEN */ +#define WM8955_TSDEN_MASK 0x0100 /* TSDEN */ +#define WM8955_TSDEN_SHIFT 8 /* TSDEN */ +#define WM8955_TSDEN_WIDTH 1 /* TSDEN */ +#define WM8955_VSEL_MASK 0x00C0 /* VSEL - [7:6] */ +#define WM8955_VSEL_SHIFT 6 /* VSEL - [7:6] */ +#define WM8955_VSEL_WIDTH 2 /* VSEL - [7:6] */ +#define WM8955_DMONOMIX_MASK 0x0030 /* DMONOMIX - [5:4] */ +#define WM8955_DMONOMIX_SHIFT 4 /* DMONOMIX - [5:4] */ +#define WM8955_DMONOMIX_WIDTH 2 /* DMONOMIX - [5:4] */ +#define WM8955_DACINV 0x0002 /* DACINV */ +#define WM8955_DACINV_MASK 0x0002 /* DACINV */ +#define WM8955_DACINV_SHIFT 1 /* DACINV */ +#define WM8955_DACINV_WIDTH 1 /* DACINV */ +#define WM8955_TOEN 0x0001 /* TOEN */ +#define WM8955_TOEN_MASK 0x0001 /* TOEN */ +#define WM8955_TOEN_SHIFT 0 /* TOEN */ +#define WM8955_TOEN_WIDTH 1 /* TOEN */ + +/* + * R24 (0x18) - Additional control (2) + */ +#define WM8955_OUT3SW_MASK 0x0180 /* OUT3SW - [8:7] */ +#define WM8955_OUT3SW_SHIFT 7 /* OUT3SW - [8:7] */ +#define WM8955_OUT3SW_WIDTH 2 /* OUT3SW - [8:7] */ +#define WM8955_ROUT2INV 0x0010 /* ROUT2INV */ +#define WM8955_ROUT2INV_MASK 0x0010 /* ROUT2INV */ +#define WM8955_ROUT2INV_SHIFT 4 /* ROUT2INV */ +#define WM8955_ROUT2INV_WIDTH 1 /* ROUT2INV */ +#define WM8955_DACOSR 0x0001 /* DACOSR */ +#define WM8955_DACOSR_MASK 0x0001 /* DACOSR */ +#define WM8955_DACOSR_SHIFT 0 /* DACOSR */ +#define WM8955_DACOSR_WIDTH 1 /* DACOSR */ + +/* + * R25 (0x19) - Power Management (1) + */ +#define WM8955_VMIDSEL_MASK 0x0180 /* VMIDSEL - [8:7] */ +#define WM8955_VMIDSEL_SHIFT 7 /* VMIDSEL - [8:7] */ +#define WM8955_VMIDSEL_WIDTH 2 /* VMIDSEL - [8:7] */ +#define WM8955_VREF 0x0040 /* VREF */ +#define WM8955_VREF_MASK 0x0040 /* VREF */ +#define WM8955_VREF_SHIFT 6 /* VREF */ +#define WM8955_VREF_WIDTH 1 /* VREF */ +#define WM8955_DIGENB 0x0001 /* DIGENB */ +#define WM8955_DIGENB_MASK 0x0001 /* DIGENB */ +#define WM8955_DIGENB_SHIFT 0 /* DIGENB */ +#define WM8955_DIGENB_WIDTH 1 /* DIGENB */ + +/* + * R26 (0x1A) - Power Management (2) + */ +#define WM8955_DACL 0x0100 /* DACL */ +#define WM8955_DACL_MASK 0x0100 /* DACL */ +#define WM8955_DACL_SHIFT 8 /* DACL */ +#define WM8955_DACL_WIDTH 1 /* DACL */ +#define WM8955_DACR 0x0080 /* DACR */ +#define WM8955_DACR_MASK 0x0080 /* DACR */ +#define WM8955_DACR_SHIFT 7 /* DACR */ +#define WM8955_DACR_WIDTH 1 /* DACR */ +#define WM8955_LOUT1 0x0040 /* LOUT1 */ +#define WM8955_LOUT1_MASK 0x0040 /* LOUT1 */ +#define WM8955_LOUT1_SHIFT 6 /* LOUT1 */ +#define WM8955_LOUT1_WIDTH 1 /* LOUT1 */ +#define WM8955_ROUT1 0x0020 /* ROUT1 */ +#define WM8955_ROUT1_MASK 0x0020 /* ROUT1 */ +#define WM8955_ROUT1_SHIFT 5 /* ROUT1 */ +#define WM8955_ROUT1_WIDTH 1 /* ROUT1 */ +#define WM8955_LOUT2 0x0010 /* LOUT2 */ +#define WM8955_LOUT2_MASK 0x0010 /* LOUT2 */ +#define WM8955_LOUT2_SHIFT 4 /* LOUT2 */ +#define WM8955_LOUT2_WIDTH 1 /* LOUT2 */ +#define WM8955_ROUT2 0x0008 /* ROUT2 */ +#define WM8955_ROUT2_MASK 0x0008 /* ROUT2 */ +#define WM8955_ROUT2_SHIFT 3 /* ROUT2 */ +#define WM8955_ROUT2_WIDTH 1 /* ROUT2 */ +#define WM8955_MONO 0x0004 /* MONO */ +#define WM8955_MONO_MASK 0x0004 /* MONO */ +#define WM8955_MONO_SHIFT 2 /* MONO */ +#define WM8955_MONO_WIDTH 1 /* MONO */ +#define WM8955_OUT3 0x0002 /* OUT3 */ +#define WM8955_OUT3_MASK 0x0002 /* OUT3 */ +#define WM8955_OUT3_SHIFT 1 /* OUT3 */ +#define WM8955_OUT3_WIDTH 1 /* OUT3 */ + +/* + * R27 (0x1B) - Additional Control (3) + */ +#define WM8955_VROI 0x0040 /* VROI */ +#define WM8955_VROI_MASK 0x0040 /* VROI */ +#define WM8955_VROI_SHIFT 6 /* VROI */ +#define WM8955_VROI_WIDTH 1 /* VROI */ + +/* + * R34 (0x22) - Left out Mix (1) + */ +#define WM8955_LD2LO 0x0100 /* LD2LO */ +#define WM8955_LD2LO_MASK 0x0100 /* LD2LO */ +#define WM8955_LD2LO_SHIFT 8 /* LD2LO */ +#define WM8955_LD2LO_WIDTH 1 /* LD2LO */ +#define WM8955_LI2LO 0x0080 /* LI2LO */ +#define WM8955_LI2LO_MASK 0x0080 /* LI2LO */ +#define WM8955_LI2LO_SHIFT 7 /* LI2LO */ +#define WM8955_LI2LO_WIDTH 1 /* LI2LO */ +#define WM8955_LI2LOVOL_MASK 0x0070 /* LI2LOVOL - [6:4] */ +#define WM8955_LI2LOVOL_SHIFT 4 /* LI2LOVOL - [6:4] */ +#define WM8955_LI2LOVOL_WIDTH 3 /* LI2LOVOL - [6:4] */ + +/* + * R35 (0x23) - Left out Mix (2) + */ +#define WM8955_RD2LO 0x0100 /* RD2LO */ +#define WM8955_RD2LO_MASK 0x0100 /* RD2LO */ +#define WM8955_RD2LO_SHIFT 8 /* RD2LO */ +#define WM8955_RD2LO_WIDTH 1 /* RD2LO */ +#define WM8955_RI2LO 0x0080 /* RI2LO */ +#define WM8955_RI2LO_MASK 0x0080 /* RI2LO */ +#define WM8955_RI2LO_SHIFT 7 /* RI2LO */ +#define WM8955_RI2LO_WIDTH 1 /* RI2LO */ +#define WM8955_RI2LOVOL_MASK 0x0070 /* RI2LOVOL - [6:4] */ +#define WM8955_RI2LOVOL_SHIFT 4 /* RI2LOVOL - [6:4] */ +#define WM8955_RI2LOVOL_WIDTH 3 /* RI2LOVOL - [6:4] */ + +/* + * R36 (0x24) - Right out Mix (1) + */ +#define WM8955_LD2RO 0x0100 /* LD2RO */ +#define WM8955_LD2RO_MASK 0x0100 /* LD2RO */ +#define WM8955_LD2RO_SHIFT 8 /* LD2RO */ +#define WM8955_LD2RO_WIDTH 1 /* LD2RO */ +#define WM8955_LI2RO 0x0080 /* LI2RO */ +#define WM8955_LI2RO_MASK 0x0080 /* LI2RO */ +#define WM8955_LI2RO_SHIFT 7 /* LI2RO */ +#define WM8955_LI2RO_WIDTH 1 /* LI2RO */ +#define WM8955_LI2ROVOL_MASK 0x0070 /* LI2ROVOL - [6:4] */ +#define WM8955_LI2ROVOL_SHIFT 4 /* LI2ROVOL - [6:4] */ +#define WM8955_LI2ROVOL_WIDTH 3 /* LI2ROVOL - [6:4] */ + +/* + * R37 (0x25) - Right Out Mix (2) + */ +#define WM8955_RD2RO 0x0100 /* RD2RO */ +#define WM8955_RD2RO_MASK 0x0100 /* RD2RO */ +#define WM8955_RD2RO_SHIFT 8 /* RD2RO */ +#define WM8955_RD2RO_WIDTH 1 /* RD2RO */ +#define WM8955_RI2RO 0x0080 /* RI2RO */ +#define WM8955_RI2RO_MASK 0x0080 /* RI2RO */ +#define WM8955_RI2RO_SHIFT 7 /* RI2RO */ +#define WM8955_RI2RO_WIDTH 1 /* RI2RO */ +#define WM8955_RI2ROVOL_MASK 0x0070 /* RI2ROVOL - [6:4] */ +#define WM8955_RI2ROVOL_SHIFT 4 /* RI2ROVOL - [6:4] */ +#define WM8955_RI2ROVOL_WIDTH 3 /* RI2ROVOL - [6:4] */ + +/* + * R38 (0x26) - Mono out Mix (1) + */ +#define WM8955_LD2MO 0x0100 /* LD2MO */ +#define WM8955_LD2MO_MASK 0x0100 /* LD2MO */ +#define WM8955_LD2MO_SHIFT 8 /* LD2MO */ +#define WM8955_LD2MO_WIDTH 1 /* LD2MO */ +#define WM8955_LI2MO 0x0080 /* LI2MO */ +#define WM8955_LI2MO_MASK 0x0080 /* LI2MO */ +#define WM8955_LI2MO_SHIFT 7 /* LI2MO */ +#define WM8955_LI2MO_WIDTH 1 /* LI2MO */ +#define WM8955_LI2MOVOL_MASK 0x0070 /* LI2MOVOL - [6:4] */ +#define WM8955_LI2MOVOL_SHIFT 4 /* LI2MOVOL - [6:4] */ +#define WM8955_LI2MOVOL_WIDTH 3 /* LI2MOVOL - [6:4] */ +#define WM8955_DMEN 0x0001 /* DMEN */ +#define WM8955_DMEN_MASK 0x0001 /* DMEN */ +#define WM8955_DMEN_SHIFT 0 /* DMEN */ +#define WM8955_DMEN_WIDTH 1 /* DMEN */ + +/* + * R39 (0x27) - Mono out Mix (2) + */ +#define WM8955_RD2MO 0x0100 /* RD2MO */ +#define WM8955_RD2MO_MASK 0x0100 /* RD2MO */ +#define WM8955_RD2MO_SHIFT 8 /* RD2MO */ +#define WM8955_RD2MO_WIDTH 1 /* RD2MO */ +#define WM8955_RI2MO 0x0080 /* RI2MO */ +#define WM8955_RI2MO_MASK 0x0080 /* RI2MO */ +#define WM8955_RI2MO_SHIFT 7 /* RI2MO */ +#define WM8955_RI2MO_WIDTH 1 /* RI2MO */ +#define WM8955_RI2MOVOL_MASK 0x0070 /* RI2MOVOL - [6:4] */ +#define WM8955_RI2MOVOL_SHIFT 4 /* RI2MOVOL - [6:4] */ +#define WM8955_RI2MOVOL_WIDTH 3 /* RI2MOVOL - [6:4] */ + +/* + * R40 (0x28) - LOUT2 volume + */ +#define WM8955_LO2VU 0x0100 /* LO2VU */ +#define WM8955_LO2VU_MASK 0x0100 /* LO2VU */ +#define WM8955_LO2VU_SHIFT 8 /* LO2VU */ +#define WM8955_LO2VU_WIDTH 1 /* LO2VU */ +#define WM8955_LO2ZC 0x0080 /* LO2ZC */ +#define WM8955_LO2ZC_MASK 0x0080 /* LO2ZC */ +#define WM8955_LO2ZC_SHIFT 7 /* LO2ZC */ +#define WM8955_LO2ZC_WIDTH 1 /* LO2ZC */ +#define WM8955_LOUT2VOL_MASK 0x007F /* LOUT2VOL - [6:0] */ +#define WM8955_LOUT2VOL_SHIFT 0 /* LOUT2VOL - [6:0] */ +#define WM8955_LOUT2VOL_WIDTH 7 /* LOUT2VOL - [6:0] */ + +/* + * R41 (0x29) - ROUT2 volume + */ +#define WM8955_RO2VU 0x0100 /* RO2VU */ +#define WM8955_RO2VU_MASK 0x0100 /* RO2VU */ +#define WM8955_RO2VU_SHIFT 8 /* RO2VU */ +#define WM8955_RO2VU_WIDTH 1 /* RO2VU */ +#define WM8955_RO2ZC 0x0080 /* RO2ZC */ +#define WM8955_RO2ZC_MASK 0x0080 /* RO2ZC */ +#define WM8955_RO2ZC_SHIFT 7 /* RO2ZC */ +#define WM8955_RO2ZC_WIDTH 1 /* RO2ZC */ +#define WM8955_ROUT2VOL_MASK 0x007F /* ROUT2VOL - [6:0] */ +#define WM8955_ROUT2VOL_SHIFT 0 /* ROUT2VOL - [6:0] */ +#define WM8955_ROUT2VOL_WIDTH 7 /* ROUT2VOL - [6:0] */ + +/* + * R42 (0x2A) - MONOOUT volume + */ +#define WM8955_MOZC 0x0080 /* MOZC */ +#define WM8955_MOZC_MASK 0x0080 /* MOZC */ +#define WM8955_MOZC_SHIFT 7 /* MOZC */ +#define WM8955_MOZC_WIDTH 1 /* MOZC */ +#define WM8955_MOUTVOL_MASK 0x007F /* MOUTVOL - [6:0] */ +#define WM8955_MOUTVOL_SHIFT 0 /* MOUTVOL - [6:0] */ +#define WM8955_MOUTVOL_WIDTH 7 /* MOUTVOL - [6:0] */ + +/* + * R43 (0x2B) - Clocking / PLL + */ +#define WM8955_MCLKSEL 0x0100 /* MCLKSEL */ +#define WM8955_MCLKSEL_MASK 0x0100 /* MCLKSEL */ +#define WM8955_MCLKSEL_SHIFT 8 /* MCLKSEL */ +#define WM8955_MCLKSEL_WIDTH 1 /* MCLKSEL */ +#define WM8955_PLLOUTDIV2 0x0020 /* PLLOUTDIV2 */ +#define WM8955_PLLOUTDIV2_MASK 0x0020 /* PLLOUTDIV2 */ +#define WM8955_PLLOUTDIV2_SHIFT 5 /* PLLOUTDIV2 */ +#define WM8955_PLLOUTDIV2_WIDTH 1 /* PLLOUTDIV2 */ +#define WM8955_PLL_RB 0x0010 /* PLL_RB */ +#define WM8955_PLL_RB_MASK 0x0010 /* PLL_RB */ +#define WM8955_PLL_RB_SHIFT 4 /* PLL_RB */ +#define WM8955_PLL_RB_WIDTH 1 /* PLL_RB */ +#define WM8955_PLLEN 0x0008 /* PLLEN */ +#define WM8955_PLLEN_MASK 0x0008 /* PLLEN */ +#define WM8955_PLLEN_SHIFT 3 /* PLLEN */ +#define WM8955_PLLEN_WIDTH 1 /* PLLEN */ + +/* + * R44 (0x2C) - PLL Control 1 + */ +#define WM8955_N_MASK 0x01E0 /* N - [8:5] */ +#define WM8955_N_SHIFT 5 /* N - [8:5] */ +#define WM8955_N_WIDTH 4 /* N - [8:5] */ +#define WM8955_K_21_18_MASK 0x000F /* K(21:18) - [3:0] */ +#define WM8955_K_21_18_SHIFT 0 /* K(21:18) - [3:0] */ +#define WM8955_K_21_18_WIDTH 4 /* K(21:18) - [3:0] */ + +/* + * R45 (0x2D) - PLL Control 2 + */ +#define WM8955_K_17_9_MASK 0x01FF /* K(17:9) - [8:0] */ +#define WM8955_K_17_9_SHIFT 0 /* K(17:9) - [8:0] */ +#define WM8955_K_17_9_WIDTH 9 /* K(17:9) - [8:0] */ + +/* + * R46 (0x2E) - PLL Control 3 + */ +#define WM8955_K_8_0_MASK 0x01FF /* K(8:0) - [8:0] */ +#define WM8955_K_8_0_SHIFT 0 /* K(8:0) - [8:0] */ +#define WM8955_K_8_0_WIDTH 9 /* K(8:0) - [8:0] */ + +/* + * R59 (0x3B) - PLL Control 4 + */ +#define WM8955_KEN 0x0080 /* KEN */ +#define WM8955_KEN_MASK 0x0080 /* KEN */ +#define WM8955_KEN_SHIFT 7 /* KEN */ +#define WM8955_KEN_WIDTH 1 /* KEN */ + +#endif -- cgit v0.10.2 From 56927eb054abd2c7371c769f359cc49a04ab488e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Dec 2009 13:11:12 +0000 Subject: ASoC: Set AIF word length for WM8904 Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 8310e5d..e44ee31 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1503,6 +1503,23 @@ static int wm8904_hw_params(struct snd_pcm_substream *substream, wm8904->bclk = snd_soc_params_to_bclk(params); } + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif1 |= 0x40; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif1 |= 0x80; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif1 |= 0xc0; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm8904->bclk); ret = wm8904_configure_clocking(codec); -- cgit v0.10.2 From 18240b67c8ca5efbbb2e8bb11942cc3db033fb16 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Dec 2009 14:20:35 +0000 Subject: ASoC: Host clock2 read up in WM8904 FLL configuration Avoids skipping over the read for disable cases. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index e44ee31..992a7f2 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1893,6 +1893,8 @@ static int wm8904_set_fll(struct snd_soc_dai *dai, int fll_id, int source, Fout == wm8904->fll_fout) return 0; + clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2); + if (Fout == 0) { dev_dbg(codec->dev, "FLL disabled\n"); @@ -1936,7 +1938,6 @@ static int wm8904_set_fll(struct snd_soc_dai *dai, int fll_id, int source, /* Save current state then disable the FLL and SYSCLK to avoid * misclocking */ - clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2); fll1 = snd_soc_read(codec, WM8904_FLL_CONTROL_1); snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2, WM8904_CLK_SYS_ENA, 0); -- cgit v0.10.2 From 0c2fd1bf4c6cb6095d8b3088d285167e66c12147 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Dec 2009 16:41:39 +0100 Subject: ALSA: hda - Check class to identify Nvidia controller chips Instead of listing all individual PCI IDs, check the matching with the PCI class together with the vendor id for Nvidia. This simplifies the pci id entries. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9b56f93..93eaf4f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2694,32 +2694,10 @@ static struct pci_device_id azx_ids[] = { /* ULI M5461 */ { PCI_DEVICE(0x10b9, 0x5461), .driver_data = AZX_DRIVER_ULI }, /* NVIDIA MCP */ - { PCI_DEVICE(0x10de, 0x026c), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0371), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x03e4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x03f0), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x044a), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0777), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x07fc), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x07fd), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac0), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be2), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(PCI_VENDOR_ID_NVIDIA, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* Creative X-Fi (CA0110-IBG) */ -- cgit v0.10.2 From ad8decb7f5dfd556e4a8400e37b127cd20d8e4c5 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 20 Dec 2009 19:01:50 +0100 Subject: ALSA: jazz16: Add support for Media Vision Jazz16 chipset This is one of Sound Blaster Pro compatible chipsets which is supported by Linux OSS driver and was missing native supoort for ALSA. The Jazz16 audio codec is Crystal CS4216 which is capable of playback and recording up to 48 kHz stereo. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai diff --git a/include/sound/sb.h b/include/sound/sb.h index 4e62ee1..9535354 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 194af3b0..755a0a5 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -239,6 +239,22 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16 card and compatibles" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB8_DSP + help + Say Y here to include support for soundcards based on the + Media Vision Jazz16 chipset: digital chip MVD1216 (Jazz16), + codec MVA416 (CS4216) and mixer MVA514 (ICS2514). + Media Vision's Jazz16 cards were sold under names Pro Sonic 16, + Premium 3-D and Pro 3-D. There were also OEMs cards with the + Jazz16 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" select SND_OPL3_LIB diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index faeffceb..af36696 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_SB8) += snd-sb8.o obj-$(CONFIG_SND_SB16) += snd-sb16.o obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o obj-$(CONFIG_SND_ES968) += snd-es968.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c new file mode 100644 index 0000000..d52966b --- /dev/null +++ b/sound/isa/sb/jazz16.c @@ -0,0 +1,385 @@ + +/* + * jazz16.c - driver for Media Vision Jazz16 based soundcards. + * Copyright (C) 2009 Krzysztof Helt + * Based on patches posted by Rask Ingemann Lambertsen and Rene Herman. + * Based on OSS Sound Blaster driver. + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive for + * more details. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include + +#define PFX "jazz16: " + +MODULE_DESCRIPTION("Media Vision Jazz16"); +MODULE_SUPPORTED_DEVICE("{{Media Vision ??? }," + "{RTL,RTL3000}}"); + +MODULE_AUTHOR("Krzysztof Helt "); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ +static unsigned long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static unsigned long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Media Vision Jazz16 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Media Vision Jazz16 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Media Vision Jazz16 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for jazz16 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for jazz16 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for jazz16 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for jazz16 driver."); +module_param_array(dma8, int, NULL, 0444); +MODULE_PARM_DESC(dma8, "DMA8 # for jazz16 driver."); +module_param_array(dma16, int, NULL, 0444); +MODULE_PARM_DESC(dma16, "DMA16 # for jazz16 driver."); + +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_DSP_GET_JAZZ_BRD_REV 0xfa +#define SB_JAZZ16_SET_DMAINTR 0xfb +#define SB_DSP_GET_JAZZ_MODEL 0xfe + +struct snd_card_jazz16 { + struct snd_sb *chip; +}; + +static irqreturn_t jazz16_interrupt(int irq, void *chip) +{ + return snd_sb8dsp_interrupt(chip); +} + +static int __devinit jazz16_configure_ports(unsigned long port, + unsigned long mpu_port, int idx) +{ + unsigned char val; + + if (!request_region(0x201, 1, "jazz16 config")) { + snd_printk(KERN_ERR "config port region is already in use.\n"); + return -EBUSY; + } + outb(SB_JAZZ16_WAKEUP - idx, 0x201); + udelay(100); + outb(SB_JAZZ16_SET_PORTS + idx, 0x201); + udelay(100); + val = port & 0x70; + val |= (mpu_port & 0x30) >> 4; + outb(val, 0x201); + + release_region(0x201, 1); + return 0; +} + +static int __devinit jazz16_detect_board(unsigned long port, + unsigned long mpu_port) +{ + int err; + int val; + struct snd_sb chip; + + if (!request_region(port, 0x10, "jazz16")) { + snd_printk(KERN_ERR "I/O port region is already in use.\n"); + return -EBUSY; + } + /* just to call snd_sbdsp_command/reset/get_byte() */ + chip.port = port; + + err = snd_sbdsp_reset(&chip); + if (err < 0) + for (val = 0; val < 4; val++) { + err = jazz16_configure_ports(port, mpu_port, val); + if (err < 0) + break; + + err = snd_sbdsp_reset(&chip); + if (!err) + break; + } + if (err < 0) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_BRD_REV)) { + err = -EBUSY; + goto err_unmap; + } + val = snd_sbdsp_get_byte(&chip); + if (val >= 0x30) + snd_sbdsp_get_byte(&chip); + + if ((val & 0xf0) != 0x10) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_MODEL)) { + err = -EBUSY; + goto err_unmap; + } + snd_sbdsp_get_byte(&chip); + err = snd_sbdsp_get_byte(&chip); + snd_printd("Media Vision Jazz16 board detected: rev 0x%x, model 0x%x\n", + val, err); + + err = 0; + +err_unmap: + release_region(port, 0x10); + return err; +} + +static int __devinit jazz16_configure_board(struct snd_sb *chip, int mpu_irq) +{ + static unsigned char jazz_irq_bits[] = { 0, 0, 2, 3, 0, 1, 0, 4, + 0, 2, 5, 0, 0, 0, 0, 6 }; + static unsigned char jazz_dma_bits[] = { 0, 1, 0, 2, 0, 3, 0, 4 }; + + if (jazz_dma_bits[chip->dma8] == 0 || + jazz_dma_bits[chip->dma16] == 0 || + jazz_irq_bits[chip->irq] == 0) + return -EINVAL; + + if (!snd_sbdsp_command(chip, SB_JAZZ16_SET_DMAINTR)) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_dma_bits[chip->dma8] | + (jazz_dma_bits[chip->dma16] << 4))) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_irq_bits[chip->irq] | + (jazz_irq_bits[mpu_irq] << 4))) + return -EBUSY; + + return 0; +} + +static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "please specify port\n"); + return 0; + } + if (dma16[dev] != SNDRV_AUTO_DMA && + dma16[dev] != 5 && dma16[dev] != 7) { + snd_printk(KERN_ERR "dma16 must be 5 or 7"); + return 0; + } + return 1; +} + +static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev) +{ + struct snd_card *card; + struct snd_card_jazz16 *jazz16; + struct snd_sb *chip; + struct snd_opl3 *opl3; + static int possible_irqs[] = {2, 3, 5, 7, 9, 10, 15, -1}; + static int possible_dmas8[] = {1, 3, -1}; + static int possible_dmas16[] = {5, 7, -1}; + int err, xirq, xdma8, xdma16, xmpu_port, xmpu_irq; + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_jazz16), &card); + if (err < 0) + return err; + + jazz16 = card->private_data; + + xirq = irq[dev]; + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + err = -EBUSY; + goto err_free; + } + } + xdma8 = dma8[dev]; + if (xdma8 == SNDRV_AUTO_DMA) { + xdma8 = snd_legacy_find_free_dma(possible_dmas8); + if (xdma8 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA8\n"); + err = -EBUSY; + goto err_free; + } + } + xdma16 = dma16[dev]; + if (xdma16 == SNDRV_AUTO_DMA) { + xdma16 = snd_legacy_find_free_dma(possible_dmas16); + if (xdma16 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA16\n"); + err = -EBUSY; + goto err_free; + } + } + + xmpu_port = mpu_port[dev]; + if (xmpu_port == SNDRV_AUTO_PORT) + xmpu_port = 0; + err = jazz16_detect_board(port[dev], xmpu_port); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 board not detected\n"); + goto err_free; + } + err = snd_sbdsp_create(card, port[dev], irq[dev], + jazz16_interrupt, + dma8[dev], dma16[dev], + SB_HW_JAZZ16, + &chip); + if (err < 0) + goto err_free; + + xmpu_irq = mpu_irq[dev]; + if (xmpu_irq == SNDRV_AUTO_IRQ || mpu_port[dev] == SNDRV_AUTO_PORT) + xmpu_irq = 0; + err = jazz16_configure_board(chip, xmpu_irq); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 configuration failed\n"); + goto err_free; + } + + jazz16->chip = chip; + + strcpy(card->driver, "jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, + "Media Vision Jazz16 at 0x%lx, irq %d, dma8 %d, dma16 %d", + port[dev], xirq, xdma8, xdma16); + + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) + goto err_free; + err = snd_sbmixer_new(chip); + if (err < 0) + goto err_free; + + err = snd_opl3_create(card, chip->port, chip->port + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) + snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", + chip->port, chip->port + 2); + else { + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto err_free; + } + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", + mpu_port[dev]); + } + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto err_free; + + dev_set_drvdata(devptr, card); + return 0; + +err_free: + snd_card_free(card); + return err; +} + +static int __devexit snd_jazz16_remove(struct device *devptr, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(devptr); + + dev_set_drvdata(devptr, NULL); + snd_card_free(card); + return 0; +} + +#ifdef CONFIG_PM +static int snd_jazz16_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *acard = card->private_data; + struct snd_sb *chip = acard->chip; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + snd_sbmixer_suspend(chip); + return 0; +} + +static int snd_jazz16_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *acard = card->private_data; + struct snd_sb *chip = acard->chip; + + snd_sbdsp_reset(chip); + snd_sbmixer_resume(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + +static struct isa_driver snd_jazz16_driver = { + .match = snd_jazz16_match, + .probe = snd_jazz16_probe, + .remove = __devexit_p(snd_jazz16_remove), +#ifdef CONFIG_PM + .suspend = snd_jazz16_suspend, + .resume = snd_jazz16_resume, +#endif + .driver = { + .name = "jazz16" + }, +}; + +static int __init alsa_card_jazz16_init(void) +{ + return isa_register_driver(&snd_jazz16_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_jazz16_exit(void) +{ + isa_unregister_driver(&snd_jazz16_driver); +} + +module_init(alsa_card_jazz16_init) +module_exit(alsa_card_jazz16_exit) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 658d557..3222aed 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -106,9 +106,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_CAPTURE_16) + return -EBUSY; + else + chip->mode |= SB_MODE_PLAYBACK_16; + } + chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -133,11 +145,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_PLAYBACK_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_PLAYBACK_8; + dma = chip->dma8; + } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); - if (runtime->channels > 1) { + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); mixreg = snd_sbmixer_read(chip, SB_DSP_STEREO_SW); @@ -147,15 +169,14 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) /* Soundblaster hardware programming reference guide, 3-23 */ snd_sbdsp_command(chip, SB_DSP_DMA8_EXIT); runtime->dma_area[0] = 0x80; - snd_dma_program(chip->dma8, runtime->dma_addr, 1, DMA_MODE_WRITE); + snd_dma_program(dma, runtime->dma_addr, 1, DMA_MODE_WRITE); /* force interrupt */ - chip->mode = SB_MODE_HALT; snd_sbdsp_command(chip, SB_DSP_OUTPUT); snd_sbdsp_command(chip, 0); snd_sbdsp_command(chip, 0); } snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save output filter status and turn it off */ @@ -168,13 +189,15 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->playback_format != SB_DSP_OUTPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -212,7 +235,6 @@ static int snd_sb8_playback_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_PLAYBACK_8 : SB_MODE_HALT; return 0; } @@ -234,9 +256,21 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_PLAYBACK_16) + return -EBUSY; + else + chip->mode |= SB_MODE_CAPTURE_16; + } + chip->capture_format = SB_DSP_LO_INPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -262,14 +296,24 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_CAPTURE_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_CAPTURE_8; + dma = chip->dma8; + } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save input filter status and turn it off */ @@ -282,13 +326,15 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->capture_format != SB_DSP_INPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -328,7 +374,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_CAPTURE_8 : SB_MODE_HALT; return 0; } @@ -339,13 +384,21 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) snd_sb_ack_8bit(chip); switch (chip->mode) { - case SB_MODE_PLAYBACK_8: /* ok.. playback is active */ + case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ + case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); break; + case SB_MODE_CAPTURE_16: + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; runtime = substream->runtime; @@ -361,10 +414,15 @@ static snd_pcm_uframes_t snd_sb8_playback_pointer(struct snd_pcm_substream *subs { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_PLAYBACK_8) + if (chip->mode & SB_MODE_PLAYBACK_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_PLAYBACK_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -372,10 +430,15 @@ static snd_pcm_uframes_t snd_sb8_capture_pointer(struct snd_pcm_substream *subst { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_CAPTURE_8) + if (chip->mode & SB_MODE_CAPTURE_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_CAPTURE_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -446,6 +509,13 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; + runtime->hw.rate_min = 4000; + runtime->hw.rate_max = 50000; + runtime->hw.channels_max = 2; + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -468,6 +538,14 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); + if (chip->dma8 > 3 || chip->dma16 >= 0) { + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 2); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 2); + runtime->hw.buffer_bytes_max = 128 * 1024 * 1024; + runtime->hw.period_bytes_max = 128 * 1024 * 1024; + } return 0; } @@ -480,6 +558,10 @@ static int snd_sb8_close(struct snd_pcm_substream *substream) chip->capture_substream = NULL; spin_lock_irqsave(&chip->open_lock, flags); chip->open &= ~SB_OPEN_PCM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + chip->mode &= ~SB_MODE_PLAYBACK; + else + chip->mode &= ~SB_MODE_CAPTURE; spin_unlock_irqrestore(&chip->open_lock, flags); return 0; } @@ -515,6 +597,7 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) struct snd_card *card = chip->card; struct snd_pcm *pcm; int err; + size_t max_prealloc = 64 * 1024; if (rpcm) *rpcm = NULL; @@ -527,9 +610,11 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops); + if (chip->dma8 > 3 || chip->dma16 >= 0) + max_prealloc = 128 * 1024; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, 64*1024); + 64*1024, max_prealloc); if (rpcm) *rpcm = pcm; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 27a6515..eae6c1c 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -170,6 +170,9 @@ static int snd_sbdsp_probe(struct snd_sb * chip) case SB_HW_CS5530: str = "16 (CS5530)"; break; + case SB_HW_JAZZ16: + str = "Pro (Jazz16)"; + break; default: return -ENODEV; } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 8cfc41f..6496822 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -779,6 +779,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -929,6 +930,7 @@ void snd_sbmixer_suspend(struct snd_sb *chip) save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -955,6 +957,7 @@ void snd_sbmixer_resume(struct snd_sb *chip) restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: -- cgit v0.10.2 From ee7c343c0134bf126b4235e65c407711b77174da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Dec 2009 12:41:37 +0100 Subject: ALSA: pcm - Add missing inclusion of linux/vmalloc.h Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index d9727c7..d6d49d6 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include -- cgit v0.10.2 From 8374e24c23448cabf6e78db2c83841c56c5df1e1 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 21 Dec 2009 17:07:08 +0100 Subject: ALSA: refine rate selection in snd_interval_ratnum() Refine the rate selection by choosing the rate closer to the requested one in case of selecting single frequency. Previously, the higher rate was always selected. Also, fix problem with the best_diff unsigned int value wrapping (turning negative). Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a27545b2..b07cc36 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -745,10 +745,13 @@ int snd_interval_ratnum(struct snd_interval *i, unsigned int rats_count, struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp) { - unsigned int best_num, best_diff, best_den; + unsigned int best_num, best_den; + int best_diff; unsigned int k; struct snd_interval t; int err; + unsigned int result_num, result_den; + int result_diff; best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { @@ -770,6 +773,8 @@ int snd_interval_ratnum(struct snd_interval *i, den -= r; } diff = num - q * den; + if (diff < 0) + diff = -diff; if (best_num == 0 || diff * best_den < best_diff * den) { best_diff = diff; @@ -784,6 +789,9 @@ int snd_interval_ratnum(struct snd_interval *i, t.min = div_down(best_num, best_den); t.openmin = !!(best_num % best_den); + result_num = best_num; + result_diff = best_diff; + result_den = best_den; best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { unsigned int num = rats[k].num; @@ -806,6 +814,8 @@ int snd_interval_ratnum(struct snd_interval *i, den += rats[k].den_step - r; } diff = q * den - num; + if (diff < 0) + diff = -diff; if (best_num == 0 || diff * best_den < best_diff * den) { best_diff = diff; @@ -825,10 +835,14 @@ int snd_interval_ratnum(struct snd_interval *i, return err; if (snd_interval_single(i)) { + if (best_diff * result_den < result_diff * best_den) { + result_num = best_num; + result_den = best_den; + } if (nump) - *nump = best_num; + *nump = result_num; if (denp) - *denp = best_den; + *denp = result_den; } return err; } -- cgit v0.10.2 From 41116e926cb92292fa4fcbe888ae8133fa0038e6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 09:00:14 +0100 Subject: ALSA: cs46xx - Fix suspend/resume with new DSP Fix the basic suspend/resume of snd-cs46xx drivers with new DSP. References: https://bugzilla.redhat.com/show_bug.cgi?id=498287 https://bugzilla.redhat.com/show_bug.cgi?id=160751 Tested-by: Florian Zumbiehl Signed-off-by: Takashi Iwai diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h index 7c44667..49b03c9 100644 --- a/include/sound/cs46xx_dsp_spos.h +++ b/include/sound/cs46xx_dsp_spos.h @@ -118,9 +118,11 @@ struct dsp_scb_descriptor { struct snd_info_entry *proc_info; int ref_count; - spinlock_t lock; - int deleted; + u16 volume[2]; + unsigned int deleted :1; + unsigned int updated :1; + unsigned int volume_set :1; }; struct dsp_task_descriptor { diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 1be96ea..e6b4a87 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -3597,7 +3597,7 @@ static struct cs_card_type __devinitdata cards[] = { #ifdef CONFIG_PM static unsigned int saved_regs[] = { BA0_ACOSV, - BA0_ASER_FADDR, + /*BA0_ASER_FADDR,*/ BA0_ASER_MASTER, BA1_PVOL, BA1_CVOL, diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index f4f0c8f..3e5ca8f 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -298,6 +298,9 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) if (ins->scbs[i].deleted) continue; cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) ); +#ifdef CONFIG_PM + kfree(ins->scbs[i].data); +#endif } kfree(ins->code.data); @@ -974,13 +977,11 @@ static struct dsp_scb_descriptor * _map_scb (struct snd_cs46xx *chip, char * nam index = find_free_scb_index (ins); + memset(&ins->scbs[index], 0, sizeof(ins->scbs[index])); strcpy(ins->scbs[index].scb_name, name); ins->scbs[index].address = dest; ins->scbs[index].index = index; - ins->scbs[index].proc_info = NULL; ins->scbs[index].ref_count = 1; - ins->scbs[index].deleted = 0; - spin_lock_init(&ins->scbs[index].lock); desc = (ins->scbs + index); ins->scbs[index].scb_symbol = add_symbol (chip, name, dest, SYMBOL_PARAMETER); @@ -1022,17 +1023,29 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) return desc; } +#define SCB_BYTES (0x10 * 4) + struct dsp_scb_descriptor * cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 dest) { struct dsp_scb_descriptor * desc; +#ifdef CONFIG_PM + /* copy the data for resume */ + scb_data = kmemdup(scb_data, SCB_BYTES, GFP_KERNEL); + if (!scb_data) + return NULL; +#endif + desc = _map_scb (chip,name,dest); if (desc) { desc->data = scb_data; _dsp_create_scb(chip,scb_data,dest); } else { snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n"); +#ifdef CONFIG_PM + kfree(scb_data); +#endif } return desc; @@ -1988,7 +2001,28 @@ int cs46xx_dsp_resume(struct snd_cs46xx * chip) continue; _dsp_create_scb(chip, s->data, s->address); } - + for (i = 0; i < ins->nscb; i++) { + struct dsp_scb_descriptor *s = &ins->scbs[i]; + if (s->deleted) + continue; + if (s->updated) + cs46xx_dsp_spos_update_scb(chip, s); + if (s->volume_set) + cs46xx_dsp_scb_set_volume(chip, s, + s->volume[0], s->volume[1]); + } + if (ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED) { + cs46xx_dsp_enable_spdif_hw(chip); + snd_cs46xx_poke(chip, (ins->ref_snoop_scb->address + 2) << 2, + (OUTPUT_SNOOP_BUFFER + 0x10) << 0x10); + if (ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) + cs46xx_poke_via_dsp(chip, SP_SPDOUT_CSUV, + ins->spdif_csuv_stream); + } + if (chip->dsp_spos_instance->spdif_status_in) { + cs46xx_poke_via_dsp(chip, SP_ASER_COUNTDOWN, 0x80000005); + cs46xx_poke_via_dsp(chip, SP_SPDIN_CONTROL, 0x800003ff); + } return 0; } #endif diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h index f9e169d..ca47a81 100644 --- a/sound/pci/cs46xx/dsp_spos.h +++ b/sound/pci/cs46xx/dsp_spos.h @@ -212,6 +212,7 @@ static inline void cs46xx_dsp_spos_update_scb (struct snd_cs46xx * chip, (scb->address + SCBsubListPtr) << 2, (scb->sub_list_ptr->address << 0x10) | (scb->next_scb_ptr->address)); + scb->updated = 1; } static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, @@ -222,6 +223,9 @@ static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl) << 2, val); snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl + 1) << 2, val); + scb->volume_set = 1; + scb->volume[0] = left; + scb->volume[1] = right; } #endif /* __DSP_SPOS_H__ */ #endif /* CONFIG_SND_CS46XX_NEW_DSP */ diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index dd7c41b..00b148a 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -115,7 +115,6 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * scb) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; - unsigned long flags; if ( scb->parent_scb_ptr ) { /* unlink parent SCB */ @@ -153,8 +152,6 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor scb->next_scb_ptr = ins->the_null_scb; } - spin_lock_irqsave(&chip->reg_lock, flags); - /* update parent first entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,scb->parent_scb_ptr); @@ -162,7 +159,6 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor cs46xx_dsp_spos_update_scb(chip,scb); scb->parent_scb_ptr = NULL; - spin_unlock_irqrestore(&chip->reg_lock, flags); } } @@ -197,9 +193,9 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * goto _end; #endif - spin_lock_irqsave(&scb->lock, flags); + spin_lock_irqsave(&chip->reg_lock, flags); _dsp_unlink_scb (chip,scb); - spin_unlock_irqrestore(&scb->lock, flags); + spin_unlock_irqrestore(&chip->reg_lock, flags); cs46xx_dsp_proc_free_scb_desc(scb); if (snd_BUG_ON(!scb->scb_symbol)) @@ -207,6 +203,10 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * remove_symbol (chip,scb->scb_symbol); ins->scbs[scb->index].deleted = 1; +#ifdef CONFIG_PM + kfree(ins->scbs[scb->index].data); + ins->scbs[scb->index].data = NULL; +#endif if (scb->index < ins->scb_highest_frag_index) ins->scb_highest_frag_index = scb->index; @@ -1508,20 +1508,17 @@ int cs46xx_dsp_pcm_unlink (struct snd_cs46xx * chip, chip->dsp_spos_instance->npcm_channels <= 0)) return -EIO; - spin_lock(&pcm_channel->src_scb->lock); - + spin_lock_irqsave(&chip->reg_lock, flags); if (pcm_channel->unlinked) { - spin_unlock(&pcm_channel->src_scb->lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); return -EIO; } - spin_lock_irqsave(&chip->reg_lock, flags); pcm_channel->unlinked = 1; - spin_unlock_irqrestore(&chip->reg_lock, flags); _dsp_unlink_scb (chip,pcm_channel->pcm_reader_scb); + spin_unlock_irqrestore(&chip->reg_lock, flags); - spin_unlock(&pcm_channel->src_scb->lock); return 0; } @@ -1533,10 +1530,10 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, struct dsp_scb_descriptor * src_scb = pcm_channel->src_scb; unsigned long flags; - spin_lock(&pcm_channel->src_scb->lock); + spin_lock_irqsave(&chip->reg_lock, flags); if (pcm_channel->unlinked == 0) { - spin_unlock(&pcm_channel->src_scb->lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); return -EIO; } @@ -1552,8 +1549,6 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, snd_BUG_ON(pcm_channel->pcm_reader_scb->parent_scb_ptr); pcm_channel->pcm_reader_scb->parent_scb_ptr = parent_scb; - spin_lock_irqsave(&chip->reg_lock, flags); - /* update SCB entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,pcm_channel->pcm_reader_scb); @@ -1562,8 +1557,6 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, pcm_channel->unlinked = 0; spin_unlock_irqrestore(&chip->reg_lock, flags); - - spin_unlock(&pcm_channel->src_scb->lock); return 0; } @@ -1596,13 +1589,17 @@ cs46xx_add_record_source (struct snd_cs46xx *chip, struct dsp_scb_descriptor * s int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src) { + unsigned long flags; + if (snd_BUG_ON(!src->parent_scb_ptr)) return -EINVAL; /* mute SCB */ cs46xx_dsp_scb_set_volume (chip,src,0,0); + spin_lock_irqsave(&chip->reg_lock, flags); _dsp_unlink_scb (chip,src); + spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; } -- cgit v0.10.2 From 75d1aeb9d6899b10420d10284e8ea894b2794224 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 11:56:32 +0100 Subject: ALSA: hda - Add Bass Speaker switch for HP dv7 The bass speaker is controlled via GPIO5. Tested-by: Wael Nasreddine Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0bafea9..a4526d0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5402,6 +5402,54 @@ static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, return 0; } +/* HP dv7 bass switch - GPIO5 */ +#define stac_hp_bass_gpio_info snd_ctl_boolean_mono_info +static int stac_hp_bass_gpio_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = !!(spec->gpio_data & 0x20); + return 0; +} + +static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_data; + + gpio_data = (spec->gpio_data & ~0x20) | + (ucontrol->value.integer.value[0] ? 0x20 : 0); + if (gpio_data == spec->gpio_data) + return 0; + spec->gpio_data = gpio_data; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); + return 1; +} + +static struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = stac_hp_bass_gpio_info, + .get = stac_hp_bass_gpio_get, + .put = stac_hp_bass_gpio_put, +}; + +static int stac_add_hp_bass_switch(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!stac_control_new(spec, &stac_hp_bass_sw_ctrl, + "Bass Speaker Playback Switch", 0)) + return -ENOMEM; + + spec->gpio_mask |= 0x20; + spec->gpio_dir |= 0x20; + spec->gpio_data |= 0x20; + return 0; +} + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5642,6 +5690,15 @@ again: return err; } + /* enable bass on HP dv7 */ + if (spec->board_config == STAC_HP_DV5) { + unsigned int cap; + cap = snd_hda_param_read(codec, 0x1, AC_PAR_GPIO_CAP); + cap &= AC_GPIO_IO_COUNT; + if (cap >= 6) + stac_add_hp_bass_switch(codec); + } + codec->proc_widget_hook = stac92hd7x_proc_hook; return 0; -- cgit v0.10.2 From 21949f00a022e090a7e8bc9a01dfca88273c6146 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Dec 2009 08:31:59 +0100 Subject: ALSA: hda - Fix NID association for capture mixers Fix the wrong implementation of NID <-> kctl mapping for capture mixers introduced by the ocmmit 5b0cb1d850c26893b1468b3a519433a1b7a176be. So far, the driver returns an error at probe. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c848ec0..29c90d7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3537,32 +3537,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) } EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); -/** - * snd_hda_add_nids - assign nids to controls from the array - * @codec: the HDA codec - * @kctl: struct snd_kcontrol - * @index: index to kctl - * @nids: the array of hda_nid_t - * @size: count of hda_nid_t items - * - * This helper function assigns NIDs in the given array to a control element. - * - * Returns 0 if successful, or a negative error code. - */ -int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, - unsigned int index, hda_nid_t *nids, unsigned int size) -{ - int err; - - for ( ; size > 0; size--, nids++) { - err = snd_hda_add_nid(codec, kctl, index, *nids); - if (err < 0) - return err; - } - return 0; -} -EXPORT_SYMBOL_HDA(snd_hda_add_nids); - #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d505d05..7cee364 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -343,8 +343,6 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); -int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, - unsigned int index, hda_nid_t *nids, unsigned int size); /* * unsolicited event handler diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 92b72d4..45ee352 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -244,8 +244,7 @@ static int ad198x_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 093cfbb..7de782a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -753,6 +753,7 @@ static int build_input(struct hda_codec *codec) spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; + int n; if (!spec->capture_bind[i]) return -ENOMEM; kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); @@ -762,10 +763,13 @@ static int build_input(struct hda_codec *codec) err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - err = snd_hda_add_nids(codec, kctl, 0, spec->adc_nid, - spec->num_inputs); - if (err < 0) - return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]); + if (err < 0) + return err; + } } if (spec->num_inputs > 1 && !spec->mic_detect) { diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index cc1c223..ff60908 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -345,8 +345,7 @@ static int cmi9880_build_controls(struct hda_codec *codec) /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->adc_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->adc_nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e7cdc6a..a451990 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2551,8 +2551,7 @@ static int alc_build_controls(struct hda_codec *codec) hda_nid_t *nids = spec->capsrc_nids; if (!nids) nids = spec->adc_nids; - err = snd_hda_add_nids(codec, kctl, i, nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index de4839e..9ddc373 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1907,8 +1907,7 @@ static int via_build_controls(struct hda_codec *codec) /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->mux_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]); if (err < 0) return err; } -- cgit v0.10.2 From 44eba3e82b35ae796826a65d8040001582adc10a Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 23 Dec 2009 18:02:41 +0100 Subject: ALSA: jazz16: refine dma and irq selection Narrow the dma and irq selection after the DOS driver. Add ALSA configuration description as well. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7a0a4a9..c540637 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1123,6 +1123,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ------------------- + + Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips: + MVD1216 + MVA416 + MVA514. + + port - port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for SB DSP chip (3,5,7,9,10,15) + dma8 - DMA # for SB DSP chip (1,3) + dma16 - DMA # for SB DSP chip (5,7) + mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) + mpu_irq - MPU-401 irq # (2,3,5,7) + + This module supports multiple cards. + Module snd-korg1212 ------------------- diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index d52966b..8d21a3f 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -189,10 +189,29 @@ static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) if (port[dev] == SNDRV_AUTO_PORT) { snd_printk(KERN_ERR "please specify port\n"); return 0; + } else if (port[dev] == 0x200 || (port[dev] & ~0x270)) { + snd_printk(KERN_ERR "incorrect port specified\n"); + return 0; + } + if (dma8[dev] != SNDRV_AUTO_DMA && + dma8[dev] != 1 && dma8[dev] != 3) { + snd_printk(KERN_ERR "dma8 must be 1 or 3\n"); + return 0; } if (dma16[dev] != SNDRV_AUTO_DMA && dma16[dev] != 5 && dma16[dev] != 7) { - snd_printk(KERN_ERR "dma16 must be 5 or 7"); + snd_printk(KERN_ERR "dma16 must be 5 or 7\n"); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x030) != 0x300) { + snd_printk(KERN_ERR "incorrect mpu_port specified\n"); + return 0; + } + if (mpu_irq[dev] != SNDRV_AUTO_DMA && + mpu_irq[dev] != 2 && mpu_irq[dev] != 3 && + mpu_irq[dev] != 5 && mpu_irq[dev] != 7) { + snd_printk(KERN_ERR "mpu_irq must be 2, 3, 5 or 7\n"); return 0; } return 1; diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 3222aed..7d84c9f 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -510,7 +510,8 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } switch (chip->hardware) { case SB_HW_JAZZ16: - runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (chip->dma16 == 5 || chip->dma16 == 7) + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; runtime->hw.rate_min = 4000; runtime->hw.rate_max = 50000; -- cgit v0.10.2 From 043958e602ac2cbf918c0dab1e4e2a7f9751ebf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 26 Dec 2009 10:36:12 +0100 Subject: ALSA: hda - Add more hints for GPIO setup of IDT/STAC codecs gpio_led, gpio_led_polarity and gpio_mute are added now. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 247be19..69dd5a4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4184,9 +4184,23 @@ static void stac_store_hints(struct hda_codec *codec) p = snd_hda_get_hint(codec, "eapd_mask"); if (p) spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "gpio_mute"); + if (p) + spec->gpio_mute = simple_strtoul(p, NULL, 0) & spec->gpio_mask; val = snd_hda_get_bool_hint(codec, "eapd_switch"); if (val >= 0) spec->eapd_switch = val; + p = snd_hda_get_hint(codec, "gpio_led_polarity"); + if (p) + spec->gpio_led_polarity = simple_strtoul(p, NULL, 0); + p = snd_hda_get_hint(codec, "gpio_led"); + if (p) { + spec->gpio_led = simple_strtoul(p, NULL, 0); + spec->gpio_mask |= spec->gpio_led; + spec->gpio_dir |= spec->gpio_led; + if (spec->gpio_led_polarity) + spec->gpio_data |= spec->gpio_led; + } } static int stac92xx_init(struct hda_codec *codec) -- cgit v0.10.2 From 92ee6162c48fab24f0676969f0f147fc12f8f21c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:18:59 +0100 Subject: ALSA: hda - Add snd_hda_shutup_pins() helper function Add a common helper function for clearing pin controls before suspend. Use the pincfg array instead of looking through all widget tree. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b3554df..94ae69f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -899,6 +899,25 @@ static void restore_pincfgs(struct hda_codec *codec) } } +/** + * snd_hda_shutup_pins - Shut up all pins + * @codec: the HDA codec + * + * Clear all pin controls to shup up before suspend for avoiding click noise. + * The controls aren't cached so that they can be resumed properly. + */ +void snd_hda_shutup_pins(struct hda_codec *codec) +{ + int i; + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + /* use read here for syncing after issuing each verb */ + snd_hda_codec_read(codec, pin->nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } +} +EXPORT_SYMBOL_HDA(snd_hda_shutup_pins); + static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0d08ad5..11c4aa8e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -898,6 +898,7 @@ int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, unsigned int cfg); int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, hda_nid_t nid, unsigned int cfg); /* for hwdep */ +void snd_hda_shutup_pins(struct hda_codec *codec); /* * Mixer diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 69dd5a4..dc1d9f1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4385,18 +4385,8 @@ static void stac92xx_free_kctls(struct hda_codec *codec) static void stac92xx_shutup(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - int i; - hda_nid_t nid; - /* reset each pin before powering down DAC/ADC to avoid click noise */ - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { - unsigned int wcaps = get_wcaps(codec, nid); - unsigned int wid_type = get_wcaps_type(wcaps); - if (wid_type == AC_WID_PIN) - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } + snd_hda_shutup_pins(codec); if (spec->eapd_mask) stac_gpio_set(codec, spec->gpio_mask, -- cgit v0.10.2 From a4e09aa3cf592d9f084ff4ceb216be40c4c265dc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:22:24 +0100 Subject: ALSA: hda - Fix click noises at suspend/free with Realtek codecs Call snd_hda_shutup_pins() at suspend and free for avoiding click noises. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6361e6b..cd6d139 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3693,6 +3693,11 @@ static int alc_build_pcms(struct hda_codec *codec) return 0; } +static inline void alc_shutup(struct hda_codec *codec) +{ + snd_hda_shutup_pins(codec); +} + static void alc_free_kctls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -3713,6 +3718,7 @@ static void alc_free(struct hda_codec *codec) if (!spec) return; + alc_shutup(codec); alc_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); @@ -3722,6 +3728,7 @@ static void alc_free(struct hda_codec *codec) static int alc_suspend(struct hda_codec *codec, pm_message_t state) { struct alc_spec *spec = codec->spec; + alc_shutup(codec); if (spec && spec->power_hook) spec->power_hook(codec, 0); return 0; -- cgit v0.10.2 From b82855a0d76ebda1cc14c00040560d77bfa042ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:24:56 +0100 Subject: ALSA: hda - Add sanity check for storing the user-defined pin configs Check whether the given NID is a pin widget before storing the user-defined pin configs. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 94ae69f..d02ea89 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -824,6 +824,9 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, struct hda_pincfg *pin; unsigned int oldcfg; + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + return -EINVAL; + oldcfg = snd_hda_codec_get_pincfg(codec, nid); pin = look_up_pincfg(codec, list, nid); if (!pin) { -- cgit v0.10.2 From 014c41fce1bd5cec381e70fc6f58fdfc96cdaf69 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 13:53:24 +0100 Subject: ALSA: hda - Use strict_strtoul() Rewrite the codes to use strict_strtoul() instead of simple_strtoul(). Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 40ccb41..b36919c 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -293,8 +293,11 @@ static ssize_t type##_store(struct device *dev, \ { \ struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ struct hda_codec *codec = hwdep->private_data; \ - char *after; \ - codec->type = simple_strtoul(buf, &after, 0); \ + unsigned long val; \ + int err = strict_strtoul(buf, 0, &val); \ + if (err < 0) \ + return err; \ + codec->type = val; \ return count; \ } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dc1d9f1..e28c810 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4159,43 +4159,47 @@ static void stac92xx_power_down(struct hda_codec *codec) static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, int enable); +static inline int get_int_hint(struct hda_codec *codec, const char *key, + int *valp) +{ + const char *p; + p = snd_hda_get_hint(codec, key); + if (p) { + unsigned long val; + if (!strict_strtoul(p, 0, &val)) { + *valp = val; + return 1; + } + } + return 0; +} + /* override some hints from the hwdep entry */ static void stac_store_hints(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - const char *p; int val; val = snd_hda_get_bool_hint(codec, "hp_detect"); if (val >= 0) spec->hp_detect = val; - p = snd_hda_get_hint(codec, "gpio_mask"); - if (p) { - spec->gpio_mask = simple_strtoul(p, NULL, 0); + if (get_int_hint(codec, "gpio_mask", &spec->gpio_mask)) { spec->eapd_mask = spec->gpio_dir = spec->gpio_data = spec->gpio_mask; } - p = snd_hda_get_hint(codec, "gpio_dir"); - if (p) - spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "gpio_data"); - if (p) - spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "eapd_mask"); - if (p) - spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "gpio_mute"); - if (p) - spec->gpio_mute = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) + spec->gpio_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) + spec->eapd_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) + spec->gpio_mute &= spec->gpio_mask; val = snd_hda_get_bool_hint(codec, "eapd_switch"); if (val >= 0) spec->eapd_switch = val; - p = snd_hda_get_hint(codec, "gpio_led_polarity"); - if (p) - spec->gpio_led_polarity = simple_strtoul(p, NULL, 0); - p = snd_hda_get_hint(codec, "gpio_led"); - if (p) { - spec->gpio_led = simple_strtoul(p, NULL, 0); + get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity); + if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { spec->gpio_mask |= spec->gpio_led; spec->gpio_dir |= spec->gpio_led; if (spec->gpio_led_polarity) -- cgit v0.10.2 From ea52bf260ecbb175339af3178c15788df21b7516 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 18:48:29 -0500 Subject: ALSA: hda: Add powerdown for Analog Devices HDA codecs This patch ports powerdown fixes to AD198x. Currently we only turn off Front and HP for suspend, but this is easily extended for additional nids. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 45ee352..cecd3c1 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -441,6 +441,11 @@ static int ad198x_build_pcms(struct hda_codec *codec) return 0; } +static inline void ad198x_shutup(struct hda_codec *codec) +{ + snd_hda_shutup_pins(codec); +} + static void ad198x_free_kctls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -454,6 +459,46 @@ static void ad198x_free_kctls(struct hda_codec *codec) snd_array_free(&spec->kctls); } +static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front, + hda_nid_t hp) +{ + struct ad198x_spec *spec = codec->spec; + snd_hda_codec_write(codec, front, 0, AC_VERB_SET_EAPD_BTLENABLE, + !spec->inv_eapd ? 0x00 : 0x02); + snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_EAPD_BTLENABLE, + !spec->inv_eapd ? 0x00 : 0x02); +} + +static void ad198x_power_eapd(struct hda_codec *codec) +{ + /* We currently only handle front, HP */ + switch (codec->vendor_id) { + case 0x11d41882: + case 0x11d4882a: + case 0x11d41884: + case 0x11d41984: + case 0x11d41883: + case 0x11d4184a: + case 0x11d4194a: + case 0x11d4194b: + ad198x_power_eapd_write(codec, 0x12, 0x11); + break; + case 0x11d41981: + case 0x11d41983: + ad198x_power_eapd_write(codec, 0x05, 0x06); + break; + case 0x11d41986: + ad198x_power_eapd_write(codec, 0x1b, 0x1a); + break; + case 0x11d41988: + case 0x11d4198b: + case 0x11d4989a: + case 0x11d4989b: + ad198x_power_eapd_write(codec, 0x29, 0x22); + break; + } +} + static void ad198x_free(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -461,11 +506,29 @@ static void ad198x_free(struct hda_codec *codec) if (!spec) return; + ad198x_shutup(codec); ad198x_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); } +#ifdef SND_HDA_NEEDS_RESUME +static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) +{ + ad198x_shutup(codec); + ad198x_power_eapd(codec); + return 0; +} + +static int ad198x_resume(struct hda_codec *codec) +{ + ad198x_init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + return 0; +} +#endif + static struct hda_codec_ops ad198x_patch_ops = { .build_controls = ad198x_build_controls, .build_pcms = ad198x_build_pcms, @@ -474,6 +537,11 @@ static struct hda_codec_ops ad198x_patch_ops = { #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = ad198x_check_power_status, #endif +#ifdef SND_HDA_NEEDS_RESUME + .suspend = ad198x_suspend, + .resume = ad198x_resume, +#endif + .reboot_notify = ad198x_shutup, }; -- cgit v0.10.2 From c97259df3f2e163c72f4d0685c61fb2e026dc989 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 18:52:08 -0500 Subject: ALSA: hda: Refactor powerdown for Realtek HDA codecs This patch converts the alc889 Aspire-specific powerdown to a generic one. Like the previous effort, it currently only handles Front and PCM but can be easily extended to cover other nids. The existing hook for alc889 Aspire-specific remains enabled. Upon further testing, I've added its use for ALC861_AUTO as well. Following patches will enable them for other quirks. Tested-by: Dr. David Alan Gilbert Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cd6d139..141ff44 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -338,7 +338,7 @@ struct alc_spec { void (*init_hook)(struct hda_codec *codec); void (*unsol_event)(struct hda_codec *codec, unsigned int res); #ifdef CONFIG_SND_HDA_POWER_SAVE - void (*power_hook)(struct hda_codec *codec, int power); + void (*power_hook)(struct hda_codec *codec); #endif /* for pin sensing */ @@ -391,7 +391,7 @@ struct alc_config_preset { void (*init_hook)(struct hda_codec *); #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_amp_list *loopbacks; - void (*power_hook)(struct hda_codec *codec, int power); + void (*power_hook)(struct hda_codec *codec); #endif }; @@ -1835,16 +1835,6 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x1b; } -#ifdef CONFIG_SND_HDA_POWER_SAVE -static void alc889_power_eapd(struct hda_codec *codec, int power) -{ - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); -} -#endif - /* * ALC880 3-stack model * @@ -3725,12 +3715,40 @@ static void alc_free(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE +static void alc_power_eapd(struct hda_codec *codec) +{ + /* We currently only handle front, HP */ + switch (codec->vendor_id) { + case 0x10ec0260: + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + break; + case 0x10ec0262: + case 0x10ec0267: + case 0x10ec0268: + case 0x10ec0269: + case 0x10ec0272: + case 0x10ec0660: + case 0x10ec0662: + case 0x10ec0663: + case 0x10ec0862: + case 0x10ec0889: + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + break; + } +} + static int alc_suspend(struct hda_codec *codec, pm_message_t state) { struct alc_spec *spec = codec->spec; alc_shutup(codec); if (spec && spec->power_hook) - spec->power_hook(codec, 0); + spec->power_hook(codec); return 0; } #endif @@ -3738,16 +3756,9 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state) #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct alc_spec *spec = codec->spec; -#endif codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (spec && spec->power_hook) - spec->power_hook(codec, 1); -#endif return 0; } #endif @@ -3767,6 +3778,7 @@ static struct hda_codec_ops alc_patch_ops = { .suspend = alc_suspend, .check_power_status = alc_check_power_status, #endif + .reboot_notify = alc_shutup, }; @@ -9547,7 +9559,7 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc889_acer_aspire_8930g_setup, .init_hook = alc_automute_amp, #ifdef CONFIG_SND_HDA_POWER_SAVE - .power_hook = alc889_power_eapd, + .power_hook = alc_power_eapd, #endif }, [ALC888_ACER_ASPIRE_7730G] = { @@ -14984,9 +14996,13 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; codec->patch_ops = alc_patch_ops; - if (board_config == ALC861_AUTO) + if (board_config == ALC861_AUTO) { spec->init_hook = alc861_auto_init; #ifdef CONFIG_SND_HDA_POWER_SAVE + spec->power_hook = alc_power_eapd; +#endif + } +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; #endif -- cgit v0.10.2 From 7d2b451e65d255427c108e990507964ac39c13ee Mon Sep 17 00:00:00 2001 From: Sergiy Kovalchuk Date: Sun, 27 Dec 2009 09:13:41 -0800 Subject: ALSA: usb-audio - Added functionality for E-mu 0404USB/0202USB/TrackerPre Added functionality: 1) Extension Units support (all XU settings now available at alsamixer, kmix, etc): - "AnalogueIn soft limiter" switch; - "Sample rate" selector (values 0,1,2,3,4,5 corresponds to 44.1 48 ... 192 kHz); - "DigitalIn CLK source" selector (internal/external) (**); - "DigitalOut format SPDIF/AC3" switch (**); (**)E-mu-0404usb only. 2) Automatic device sample rate adjustment depending on substream samplerate for both capture and playback substream. [minor coding-style fixes by tiwai] Signed-off-by: Sergiy Kovalchuk Signed-off-by: Takashi Iwai diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 31b63ea..286fa14 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1271,6 +1271,47 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, } /* + * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * not for interface. + */ +static void set_format_emu_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt) +{ + unsigned char emu_samplerate_id = 0; + + /* When capture is active + * sample rate shouldn't be changed + * by playback substream + */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (subs->stream->substream[SNDRV_PCM_STREAM_CAPTURE].interface != -1) + return; + } + + switch (fmt->rate_min) { + case 48000: + emu_samplerate_id = EMU_QUIRK_SR_48000HZ; + break; + case 88200: + emu_samplerate_id = EMU_QUIRK_SR_88200HZ; + break; + case 96000: + emu_samplerate_id = EMU_QUIRK_SR_96000HZ; + break; + case 176400: + emu_samplerate_id = EMU_QUIRK_SR_176400HZ; + break; + case 192000: + emu_samplerate_id = EMU_QUIRK_SR_192000HZ; + break; + default: + emu_samplerate_id = EMU_QUIRK_SR_44100HZ; + break; + } + snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); +} + +/* * find a matching format and set up the interface */ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) @@ -1383,6 +1424,14 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->cur_audiofmt = fmt; + switch (subs->stream->chip->usb_id) { + case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ + case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ + case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + set_format_emu_quirk(subs, fmt); + break; + } + #if 0 printk(KERN_DEBUG "setting done: format = %d, rate = %d..%d, channels = %d\n", diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9826337..1522167 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -208,6 +208,16 @@ struct snd_usb_midi_endpoint_info { /* */ +/*E-mu USB samplerate control quirk*/ +enum { + EMU_QUIRK_SR_44100HZ = 0, + EMU_QUIRK_SR_48000HZ, + EMU_QUIRK_SR_88200HZ, + EMU_QUIRK_SR_96000HZ, + EMU_QUIRK_SR_176400HZ, + EMU_QUIRK_SR_192000HZ +}; + #define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8)) #define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16)) #define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24)) @@ -233,6 +243,9 @@ void snd_usbmidi_input_stop(struct list_head* p); void snd_usbmidi_input_start(struct list_head* p); void snd_usbmidi_disconnect(struct list_head *p); +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, + unsigned char samplerate_id); + /* * retrieve usb_interface descriptor from the host interface * (conditional for compatibility with the older API) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index c998220..f5596cf 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -186,6 +186,21 @@ enum { USB_PROC_DCR_RELEASE = 6, }; +/*E-mu 0202(0404) eXtension Unit(XU) control*/ +enum { + USB_XU_CLOCK_RATE = 0xe301, + USB_XU_CLOCK_SOURCE = 0xe302, + USB_XU_DIGITAL_IO_STATUS = 0xe303, + USB_XU_DEVICE_OPTIONS = 0xe304, + USB_XU_DIRECT_MONITORING = 0xe305, + USB_XU_METERING = 0xe306 +}; +enum { + USB_XU_CLOCK_SOURCE_SELECTOR = 0x02, /* clock source*/ + USB_XU_CLOCK_RATE_SELECTOR = 0x03, /* clock rate */ + USB_XU_DIGITAL_FORMAT_SELECTOR = 0x01, /* the spdif format */ + USB_XU_SOFT_LIMIT_SELECTOR = 0x03 /* soft limiter */ +}; /* * manual mapping of mixer names @@ -1330,7 +1345,32 @@ static struct procunit_info procunits[] = { { USB_PROC_DCR, "DCR", dcr_proc_info }, { 0 }, }; - +/* + * predefined data for extension units + */ +static struct procunit_value_info clock_rate_xu_info[] = { + { USB_XU_CLOCK_RATE_SELECTOR, "Selector", USB_MIXER_U8, 0 }, + { 0 } +}; +static struct procunit_value_info clock_source_xu_info[] = { + { USB_XU_CLOCK_SOURCE_SELECTOR, "External", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_value_info spdif_format_xu_info[] = { + { USB_XU_DIGITAL_FORMAT_SELECTOR, "SPDIF/AC3", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_value_info soft_limit_xu_info[] = { + { USB_XU_SOFT_LIMIT_SELECTOR, " ", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_info extunits[] = { + { USB_XU_CLOCK_RATE, "Clock rate", clock_rate_xu_info }, + { USB_XU_CLOCK_SOURCE, "DigitalIn CLK source", clock_source_xu_info }, + { USB_XU_DIGITAL_IO_STATUS, "DigitalOut format:", spdif_format_xu_info }, + { USB_XU_DEVICE_OPTIONS, "AnalogueIn Soft Limit", soft_limit_xu_info }, + { 0 } +}; /* * build a processing/extension unit */ @@ -1391,8 +1431,18 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned cval->max = dsc[15]; cval->res = 1; cval->initialized = 1; - } else - get_min_max(cval, valinfo->min_value); + } else { + if (type == USB_XU_CLOCK_RATE) { + /* E-Mu USB 0404/0202/TrackerPre + * samplerate control quirk + */ + cval->min = 0; + cval->max = 5; + cval->res = 1; + cval->initialized = 1; + } else + get_min_max(cval, valinfo->min_value); + } kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (! kctl) { @@ -1433,7 +1483,7 @@ static int parse_audio_processing_unit(struct mixer_build *state, int unitid, un static int parse_audio_extension_unit(struct mixer_build *state, int unitid, unsigned char *desc) { - return build_audio_procunit(state, unitid, desc, NULL, "Extension Unit"); + return build_audio_procunit(state, unitid, desc, extunits, "Extension Unit"); } @@ -2109,6 +2159,23 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) return 0; } +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, + unsigned char samplerate_id) +{ + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid = 12; /* SamleRate ExtensionUnit ID */ + + list_for_each_entry(mixer, &chip->mixer_list, list) { + cval = mixer->id_elems[unitid]; + if (cval) { + set_cur_ctl_value(cval, cval->control << 8, samplerate_id); + snd_usb_mixer_notify_id(mixer, unitid); + } + break; + } +} + int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error) { -- cgit v0.10.2 From adc8d31326c32a2a1e145ab80accbc3c6570b117 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 27 Dec 2009 12:19:57 -0500 Subject: ALSA: usb-audio: make buffer pointer based on bytes instead on frames Since there are devices that do not align the size of their data packets to frame boundaries, the driver needs to be able to keep track of partial frames. This patch prepares for support for such devices by changing the hwptr_done variable from a frame counter to a byte counter. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 286fa14..8fcb5d5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -173,7 +173,7 @@ struct snd_usb_substream { unsigned int running: 1; /* running status */ - unsigned int hwptr_done; /* processed frame position in the buffer */ + unsigned int hwptr_done; /* processed byte position in the buffer */ unsigned int transfer_done; /* processed frames since last period update */ unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ @@ -342,7 +342,7 @@ static int retire_capture_urb(struct snd_usb_substream *subs, unsigned long flags; unsigned char *cp; int i; - unsigned int stride, len, oldptr; + unsigned int stride, frames, bytes, oldptr; int period_elapsed = 0; stride = runtime->frame_bits >> 3; @@ -353,29 +353,28 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - len = urb->iso_frame_desc[i].actual_length / stride; - if (! len) - continue; + frames = urb->iso_frame_desc[i].actual_length / stride; + bytes = frames * stride; /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); oldptr = subs->hwptr_done; - subs->hwptr_done += len; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; - subs->transfer_done += len; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + subs->transfer_done += frames; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; } spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ - if (oldptr + len > runtime->buffer_size) { - unsigned int cnt = runtime->buffer_size - oldptr; - unsigned int blen = cnt * stride; - memcpy(runtime->dma_area + oldptr * stride, cp, blen); - memcpy(runtime->dma_area, cp + blen, len * stride - blen); + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); } else { - memcpy(runtime->dma_area + oldptr * stride, cp, len * stride); + memcpy(runtime->dma_area + oldptr, cp, bytes); } } if (period_elapsed) @@ -562,24 +561,24 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { - int i, stride, offs; - unsigned int counts; + int i, stride; + unsigned int counts, frames, bytes; unsigned long flags; int period_elapsed = 0; struct snd_urb_ctx *ctx = urb->context; stride = runtime->frame_bits >> 3; - offs = 0; + frames = 0; urb->dev = ctx->subs->dev; /* we need to set this at each time */ urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); for (i = 0; i < ctx->packets; i++) { counts = snd_usb_audio_next_packet_size(subs); /* set up descriptor */ - urb->iso_frame_desc[i].offset = offs * stride; + urb->iso_frame_desc[i].offset = frames * stride; urb->iso_frame_desc[i].length = counts * stride; - offs += counts; + frames += counts; urb->number_of_packets++; subs->transfer_done += counts; if (subs->transfer_done >= runtime->period_size) { @@ -589,7 +588,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not * supported yet */ - offs -= subs->transfer_done; + frames -= subs->transfer_done; counts -= subs->transfer_done; urb->iso_frame_desc[i].length = counts * stride; @@ -599,7 +598,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (i < ctx->packets) { /* add a transfer delimiter */ urb->iso_frame_desc[i].offset = - offs * stride; + frames * stride; urb->iso_frame_desc[i].length = 0; urb->number_of_packets++; } @@ -609,26 +608,25 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (period_elapsed) /* finish at the period boundary */ break; } - if (subs->hwptr_done + offs > runtime->buffer_size) { + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { /* err, the transferred area goes over buffer boundary. */ - unsigned int len = runtime->buffer_size - subs->hwptr_done; + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done * stride, - len * stride); - memcpy(urb->transfer_buffer + len * stride, - runtime->dma_area, - (offs - len) * stride); + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); } else { memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done * stride, - offs * stride); + runtime->dma_area + subs->hwptr_done, bytes); } - subs->hwptr_done += offs; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; - runtime->delay += offs; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + runtime->delay += frames; spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = offs * stride; + urb->transfer_buffer_length = bytes; if (period_elapsed) snd_pcm_period_elapsed(subs->pcm_substream); return 0; @@ -901,18 +899,18 @@ static int wait_clear_urbs(struct snd_usb_substream *subs) /* - * return the current pcm pointer. just return the hwptr_done value. + * return the current pcm pointer. just based on the hwptr_done value. */ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs; - snd_pcm_uframes_t hwptr_done; + unsigned int hwptr_done; subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; spin_unlock(&subs->lock); - return hwptr_done; + return hwptr_done / (substream->runtime->frame_bits >> 3); } -- cgit v0.10.2 From 98e89f606c38a310a20342f90e0c453e6afadf18 Mon Sep 17 00:00:00 2001 From: "John S. Gruber" Date: Sun, 27 Dec 2009 12:19:58 -0500 Subject: ALSA: usb-audio: relax urb data align. restriction HVR-950Q and HVR-850 only Addressing audio quality problem. In sound/usb/usbaudio.c, for the Hauppage HVR-950Q and HVR-850 only, change retire_capture_urb to allow transfers on audio sub-slot boundaries rather than audio slots boundaries. With these devices the left and right channel samples can be split between two different urbs. Throwing away extra channel samples causes a sound quality problem for stereo streams as the left and right channels are swapped repeatedly, perhaps many times per second. Urbs unaligned on sub-slot boundaries are still truncated to the next lowest stride (audio slot) to retain synchronization on samples even though left/right channel synchronization may be lost in this case. Detect the quirk using a case statement in snd_usb_audio_probe. BugLink: https://bugs.launchpad.net/ubuntu/+bug/495745 Signed-off-by: John S. Gruber Signed-off-by: Takashi Iwai diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8fcb5d5..617515f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -169,6 +169,7 @@ struct snd_usb_substream { unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int fill_max: 1; /* fill max packet size always */ + unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int running: 1; /* running status */ @@ -353,14 +354,25 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - frames = urb->iso_frame_desc[i].actual_length / stride; - bytes = frames * stride; + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); oldptr = subs->hwptr_done; subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; subs->transfer_done += frames; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; @@ -2238,6 +2250,7 @@ static void init_substream(struct snd_usb_stream *as, int stream, struct audiofo subs->stream = as; subs->direction = stream; subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { subs->ops = audio_urb_ops[stream]; } else { @@ -3618,6 +3631,20 @@ static void *snd_usb_audio_probe(struct usb_device *dev, } } + switch (chip->usb_id) { + case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ + case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ + chip->txfr_quirk = 1; + break; + default: + chip->txfr_quirk = 0; + } err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1522167..d180554b8 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -125,6 +125,7 @@ struct snd_usb_audio { struct snd_card *card; u32 usb_id; int shutdown; + unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ int num_interfaces; int num_suspended_intf; -- cgit v0.10.2 From 52a7a5835173af61b9f6c3038212370d9717526f Mon Sep 17 00:00:00 2001 From: "John S. Gruber" Date: Sun, 27 Dec 2009 12:19:59 -0500 Subject: ALSA: usb-audio: use usbquirk.h for detection of HVR-950Q/850 Detect the HVR-950Q HVR-850 urb data alignment quirk using usbquirk.h rather than using a case statement in snd_usb_audio_probe. Signed-off-by: John S. Gruber Signed-off-by: Takashi Iwai diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 617515f..4ada98e 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3203,6 +3203,18 @@ static int ignore_interface_quirk(struct snd_usb_audio *chip, return 0; } +/* + * Allow alignment on audio sub-slot (channel samples) rather than + * on audio slots (audio frames) + */ +static int create_align_transfer_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) +{ + chip->txfr_quirk = 1; + return 1; /* Continue with creating streams and mixer */ +} + /* * boot quirks @@ -3377,7 +3389,8 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, - [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, + [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { @@ -3631,20 +3644,7 @@ static void *snd_usb_audio_probe(struct usb_device *dev, } } - switch (chip->usb_id) { - case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ - case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ - chip->txfr_quirk = 1; - break; - default: - chip->txfr_quirk = 0; - } + chip->txfr_quirk = 0; err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d180554b8..9d8cea4 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -161,6 +161,7 @@ enum quirk_type { QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UAXX, + QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index bd6706c..65bbd22 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2074,6 +2074,120 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Hauppauge HVR-950Q and HVR-850 */ +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7200), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7201), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7202), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7203), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7204), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7205), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7250), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7230), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-850", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- cgit v0.10.2 From 4757968dbff3d43f373f08de973014a9bd41ef0a Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 28 Dec 2009 16:15:03 +0100 Subject: ALSA: Release v1.0.22.1 Signed-off-by: Jaroslav Kysela diff --git a/include/sound/version.h b/include/sound/version.h index 1f5d4872..7fed234 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.22" +#define CONFIG_SND_VERSION "1.0.22.1" #define CONFIG_SND_DATE "" -- cgit v0.10.2 From afe1c2cd71eb4e0fade720b5709722e7124f29c0 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Fri, 25 Dec 2009 14:10:06 +0800 Subject: ASoC: ad1836: reset and restore clock control mode in suspend/resume entry Tests show frequent suspend/resume(frequent poweroff/on ad1836 internal components) maybe make ad1836 clock mode wrong sometimes after wakeup. This patch reset/restore ad1836 clock mode while executing PM, then ad1836 can always resume to right clock status. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 2c18e3d..83add2f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,6 +223,36 @@ static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, return reg_cache[reg]; } +#ifdef CONFIG_PM +static int ad1836_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* reset clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} + +static int ad1836_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* restore clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 |= AD1836_ADC_AUX; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} +#else +#define ad1836_soc_suspend NULL +#define ad1836_soc_resume NULL +#endif + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -404,6 +434,8 @@ static int ad1836_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, + .suspend = ad1836_soc_suspend, + .resume = ad1836_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); -- cgit v0.10.2 From 08ba864e2789a94c259b8d0aee13a5a183edd46e Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Fri, 25 Dec 2009 14:10:07 +0800 Subject: ASoC: ad1938: fix typo, rename mask to rx_mask for ad1938_set_tdm_slot Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 5d48918..735c356 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -145,7 +145,7 @@ static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd) } static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, - unsigned int mask, int slots, int width) + unsigned int rx_mask, int slots, int width) { struct snd_soc_codec *codec = dai->codec; int dac_reg = codec->read(codec, AD1938_DAC_CTRL1); -- cgit v0.10.2 From 5b61735534193ab357636d5b56c098f0bbe8bac8 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Fri, 25 Dec 2009 14:10:08 +0800 Subject: ASoC: ad1938: let soc-core dapm handle PLL power PM architecture of ad1938 is simple, we don't need a bundle of functions like ad1938_pll_powerctrl, ad1938_set_bias_level for only PLL. A dapm supply will handle on/off of PLL. Since soc-core can poweron/off PLL on-demand, we don't need to poweron/off PLL in suspend/resume entries too. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 735c356..47d9ac0 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -97,6 +97,7 @@ static const struct snd_kcontrol_new ad1938_snd_controls[] = { static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("PLL_PWR", AD1938_PLL_CLK_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), @@ -107,6 +108,8 @@ static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { }; static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "PLL_PWR" }, + { "ADC", NULL, "PLL_PWR" }, { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", "DAC1 Switch", "DAC" }, @@ -134,16 +137,6 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) return 0; } -static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd) -{ - int reg = codec->read(codec, AD1938_PLL_CLK_CTRL0); - reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | - AD1938_PLL_POWERDOWN; - codec->write(codec, AD1938_PLL_CLK_CTRL0, reg); - - return 0; -} - static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int width) { @@ -306,24 +299,6 @@ static int ad1938_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ad1938_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - switch (level) { - case SND_SOC_BIAS_ON: - ad1938_pll_powerctrl(codec, 1); - break; - case SND_SOC_BIAS_PREPARE: - break; - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - ad1938_pll_powerctrl(codec, 0); - break; - } - codec->bias_level = level; - return 0; -} - /* * interface to read/write ad1938 register */ @@ -514,7 +489,6 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->num_dai = 1; codec->write = ad1938_write_reg; codec->read = ad1938_read_reg_cache; - codec->set_bias_level = ad1938_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -559,7 +533,6 @@ static int ad1938_register(struct ad1938_priv *ad1938) static void ad1938_unregister(struct ad1938_priv *ad1938) { - ad1938_set_bias_level(&ad1938->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dai(&ad1938_dai); snd_soc_unregister_codec(&ad1938->codec); kfree(ad1938); @@ -593,7 +566,6 @@ static int ad1938_probe(struct platform_device *pdev) ARRAY_SIZE(ad1938_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY); pcm_err: return ret; @@ -610,37 +582,9 @@ static int ad1938_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM -static int ad1938_suspend(struct platform_device *pdev, - pm_message_t state) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - ad1938_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int ad1938_resume(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->suspend_bias_level == SND_SOC_BIAS_ON) - ad1938_set_bias_level(codec, SND_SOC_BIAS_ON); - - return 0; -} -#else -#define ad1938_suspend NULL -#define ad1938_resume NULL -#endif - struct snd_soc_codec_device soc_codec_dev_ad1938 = { .probe = ad1938_probe, .remove = ad1938_remove, - .suspend = ad1938_suspend, - .resume = ad1938_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1938); -- cgit v0.10.2 From 1c418d1f623438147a485db987de296ab372e0f3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 28 Dec 2009 14:09:05 +0900 Subject: ASoC: fsi: Add over_period flag to prevent the misunderstanding Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7506ef6..b311a9e 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -373,14 +373,16 @@ static int fsi_data_push(struct fsi_priv *fsi) int fifo_free; int width; u8 *start; - int i; + int i, over_period; if (!fsi || !fsi->substream || !fsi->substream->runtime) return -EINVAL; - runtime = fsi->substream->runtime; + over_period = 0; + substream = fsi->substream; + runtime = substream->runtime; /* FSI FIFO has limit. * So, this driver can not send periods data at a time @@ -388,7 +390,7 @@ static int fsi_data_push(struct fsi_priv *fsi) if (fsi->byte_offset >= fsi->period_len * (fsi->periods + 1)) { - substream = fsi->substream; + over_period = 1; fsi->periods = (fsi->periods + 1) % runtime->periods; if (0 == fsi->periods) @@ -429,7 +431,7 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi_irq_enable(fsi, 1); - if (substream) + if (over_period) snd_pcm_period_elapsed(substream); return 0; @@ -443,14 +445,16 @@ static int fsi_data_pop(struct fsi_priv *fsi) int fifo_fill; int width; u8 *start; - int i; + int i, over_period; if (!fsi || !fsi->substream || !fsi->substream->runtime) return -EINVAL; - runtime = fsi->substream->runtime; + over_period = 0; + substream = fsi->substream; + runtime = substream->runtime; /* FSI FIFO has limit. * So, this driver can not send periods data at a time @@ -458,7 +462,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) if (fsi->byte_offset >= fsi->period_len * (fsi->periods + 1)) { - substream = fsi->substream; + over_period = 1; fsi->periods = (fsi->periods + 1) % runtime->periods; if (0 == fsi->periods) @@ -498,7 +502,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi_irq_enable(fsi, 0); - if (substream) + if (over_period) snd_pcm_period_elapsed(substream); return 0; -- cgit v0.10.2 From 142e8174b3c493f40469d3ecee0e404645e9c483 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 28 Dec 2009 14:09:11 +0900 Subject: ASoC: fsi: Add fsi_get_dai to get snd_soc_dai Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b311a9e..d078151 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -210,11 +210,17 @@ static int fsi_is_port_a(struct fsi_priv *fsi) return fsi->master->base == fsi->base; } -static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_dai *dai = machine->cpu_dai; + + return machine->cpu_dai; +} + +static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +{ + struct snd_soc_dai *dai = fsi_get_dai(substream); return dai->private_data; } -- cgit v0.10.2 From 59c3b003ddd3c815de1aa015920710a9e4bf195b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 28 Dec 2009 14:09:16 +0900 Subject: ASoC: fsi: Add over/under run error settlement Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index d078151..123cd6f 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -67,6 +67,7 @@ /* DOFF_ST */ #define ERR_OVER 0x00000010 #define ERR_UNDER 0x00000001 +#define ST_ERR (ERR_OVER | ERR_UNDER) /* CLK_RST */ #define B_CLK 0x00000010 @@ -375,11 +376,12 @@ static int fsi_data_push(struct fsi_priv *fsi) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; + u32 status; int send; int fifo_free; int width; u8 *start; - int i, over_period; + int i, ret, over_period; if (!fsi || !fsi->substream || @@ -435,23 +437,33 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi->byte_offset += send * width; + ret = 0; + status = fsi_reg_read(fsi, DOFF_ST); + if (status & ERR_OVER) { + struct snd_soc_dai *dai = fsi_get_dai(substream); + dev_err(dai->dev, "over run error\n"); + fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR); + ret = -EIO; + } + fsi_irq_enable(fsi, 1); if (over_period) snd_pcm_period_elapsed(substream); - return 0; + return ret; } static int fsi_data_pop(struct fsi_priv *fsi) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; + u32 status; int free; int fifo_fill; int width; u8 *start; - int i, over_period; + int i, ret, over_period; if (!fsi || !fsi->substream || @@ -506,12 +518,21 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi->byte_offset += fifo_fill * width; + ret = 0; + status = fsi_reg_read(fsi, DIFF_ST); + if (status & ERR_UNDER) { + struct snd_soc_dai *dai = fsi_get_dai(substream); + dev_err(dai->dev, "under run error\n"); + fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR); + ret = -EIO; + } + fsi_irq_enable(fsi, 0); if (over_period) snd_pcm_period_elapsed(substream); - return 0; + return ret; } static irqreturn_t fsi_interrupt(int irq, void *data) -- cgit v0.10.2 From 8998c89907f84f7e25536c1c670a134c831e682f Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Thu, 31 Dec 2009 10:30:34 +0800 Subject: ASoC: soc-cache: cleanup training whitespace and coding style Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d2505e8..02c2357 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -182,7 +182,7 @@ static struct { { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, - .spi_write = snd_soc_7_9_spi_write + .spi_write = snd_soc_7_9_spi_write, }, { .addr_bits = 8, .data_bits = 8, -- cgit v0.10.2 From 7427b4b9a63fd7e051d642ff0f12ef8337c08bb3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:19 +0200 Subject: ASoC: tlv320dac33: Change nsample switch to FIFO mode enum In order to have support for more FIFO modes supported by tlv320dac33, the switch for enabling/disabling the FIFO use has to be replaced with an enum. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 5037454..b67961dd 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -59,6 +59,12 @@ enum dac33_state { DAC33_FLUSH, }; +enum dac33_fifo_modes { + DAC33_FIFO_BYPASS = 0, + DAC33_FIFO_MODE1, + DAC33_FIFO_LAST_MODE, +}; + #define DAC33_NUM_SUPPLIES 3 static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { "AVDD", @@ -82,7 +88,7 @@ struct tlv320dac33_priv { * this */ unsigned int nsample_max; /* nsample should not be higher than * this */ - unsigned int nsample_switch; /* Use FIFO or bypass FIFO switch */ + enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */ unsigned int nsample; /* burst read amount from host */ enum dac33_state state; @@ -381,39 +387,48 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol, return ret; } -static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol, +static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct tlv320dac33_priv *dac33 = codec->private_data; - ucontrol->value.integer.value[0] = dac33->nsample_switch; + ucontrol->value.integer.value[0] = dac33->fifo_mode; return 0; } -static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol, +static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct tlv320dac33_priv *dac33 = codec->private_data; int ret = 0; - if (dac33->nsample_switch == ucontrol->value.integer.value[0]) + if (dac33->fifo_mode == ucontrol->value.integer.value[0]) return 0; /* Do not allow changes while stream is running*/ if (codec->active) return -EPERM; if (ucontrol->value.integer.value[0] < 0 || - ucontrol->value.integer.value[0] > 1) + ucontrol->value.integer.value[0] >= DAC33_FIFO_LAST_MODE) ret = -EINVAL; else - dac33->nsample_switch = ucontrol->value.integer.value[0]; + dac33->fifo_mode = ucontrol->value.integer.value[0]; return ret; } +/* Codec operation modes */ +static const char *dac33_fifo_mode_texts[] = { + "Bypass", "Mode 1" +}; + +static const struct soc_enum dac33_fifo_mode_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts), + dac33_fifo_mode_texts); + /* * DACL/R digital volume control: * from 0 dB to -63.5 in 0.5 dB steps @@ -436,8 +451,8 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = { static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, dac33_get_nsample, dac33_set_nsample), - SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0, - dac33_get_nsample_switch, dac33_set_nsample_switch), + SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum, + dac33_get_fifo_mode, dac33_set_fifo_mode), }; /* Analog bypass */ @@ -586,7 +601,7 @@ static void dac33_shutdown(struct snd_pcm_substream *substream, unsigned int pwr_ctrl; /* Stop pending workqueue */ - if (dac33->nsample_switch) + if (dac33->fifo_mode) cancel_work_sync(&dac33->work); mutex_lock(&dac33->mutex); @@ -714,7 +729,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_oscwait(codec); - if (dac33->nsample_switch) { + if (dac33->fifo_mode) { /* 50-51 : ASRC Control registers */ dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ @@ -734,7 +749,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ } - if (dac33->nsample_switch) + if (dac33->fifo_mode) fifoctrl_a &= ~DAC33_FBYPAS; else fifoctrl_a |= DAC33_FBYPAS; @@ -742,13 +757,13 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); - if (dac33->nsample_switch) + if (dac33->fifo_mode) reg_tmp &= ~DAC33_BCLKON; else reg_tmp |= DAC33_BCLKON; dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); - if (dac33->nsample_switch) { + if (dac33->fifo_mode) { /* 20: BCLK divide ratio */ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); @@ -828,7 +843,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (dac33->nsample_switch) { + if (dac33->fifo_mode) { dac33->state = DAC33_PREFILL; queue_work(dac33->dac33_wq, &dac33->work); } @@ -836,7 +851,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (dac33->nsample_switch) { + if (dac33->fifo_mode) { dac33->state = DAC33_FLUSH; queue_work(dac33->dac33_wq, &dac33->work); } @@ -1125,7 +1140,7 @@ static int dac33_i2c_probe(struct i2c_client *client, dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; /* Disable FIFO use by default */ - dac33->nsample_switch = 0; + dac33->fifo_mode = DAC33_FIFO_BYPASS; tlv320dac33_codec = codec; -- cgit v0.10.2 From d4f102d437c069a64f3a4c7a6cd50360e034541f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:20 +0200 Subject: ASoC: tlv320dac33: Introduce prefill and playback state handlers Ensure that the code is going to be readable, when new FIFO modes are introduced later. Move the prefill and playback state handling to inlined functions. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index b67961dd..f7c7bbc 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -543,6 +543,44 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, return 0; } +static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) +{ + struct snd_soc_codec *codec; + + codec = &dac33->codec; + + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + dac33_write16(codec, DAC33_PREFILL_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + break; + default: + dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", + dac33->fifo_mode); + break; + } +} + +static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) +{ + struct snd_soc_codec *codec; + + codec = &dac33->codec; + + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + break; + default: + dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", + dac33->fifo_mode); + break; + } +} + static void dac33_work(struct work_struct *work) { struct snd_soc_codec *codec; @@ -556,14 +594,10 @@ static void dac33_work(struct work_struct *work) switch (dac33->state) { case DAC33_PREFILL: dac33->state = DAC33_PLAYBACK; - dac33_write16(codec, DAC33_NSAMPLE_MSB, - DAC33_THRREG(dac33->nsample)); - dac33_write16(codec, DAC33_PREFILL_MSB, - DAC33_THRREG(dac33->alarm_threshold)); + dac33_prefill_handler(dac33); break; case DAC33_PLAYBACK: - dac33_write16(codec, DAC33_NSAMPLE_MSB, - DAC33_THRREG(dac33->nsample)); + dac33_playback_handler(dac33); break; case DAC33_IDLE: break; -- cgit v0.10.2 From aec242dc3719e19bd7c1561f8a56a4eb37bb3987 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:21 +0200 Subject: ASoC: tlv320dac33: Clean up the hardware configuration code Use switch instead of if statements to configure FIFO bypass and mode1. With this change adding new FIFO mode is going to be easier, and cleaner. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f7c7bbc..c684aa2 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -707,7 +707,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) struct snd_soc_codec *codec = socdev->card->codec; struct tlv320dac33_priv *dac33 = codec->private_data; unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; - u8 aictrl_a, fifoctrl_a; + u8 aictrl_a, aictrl_b, fifoctrl_a; switch (substream->runtime->rate) { case 44100: @@ -764,6 +764,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_oscwait(codec); if (dac33->fifo_mode) { + /* Generic for all FIFO modes */ /* 50-51 : ASRC Control registers */ dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ @@ -773,38 +774,66 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) /* Set interrupts to high active */ dac33_write(codec, DAC33_INTP_CTRL_A, DAC33_INTPM_AHIGH); - - dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, - DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); - dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); } else { + /* FIFO bypass mode */ /* 50-51 : ASRC Control registers */ dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCBYP); dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ } - if (dac33->fifo_mode) + /* Interrupt behaviour configuration */ + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: + dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, + DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); + dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); + break; + default: + /* in FIFO bypass mode, the interrupts are not used */ + break; + } + + aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: + /* + * For mode1: + * Disable the FIFO bypass (Enable the use of FIFO) + * Select nSample mode + * BCLK is only running when data is needed by DAC33 + */ fifoctrl_a &= ~DAC33_FBYPAS; - else + fifoctrl_a &= ~DAC33_FAUTO; + aictrl_b &= ~DAC33_BCLKON; + break; + default: + /* + * For FIFO bypass mode: + * Enable the FIFO bypass (Disable the FIFO use) + * Set the BCLK as continous + */ fifoctrl_a |= DAC33_FBYPAS; - dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); + aictrl_b |= DAC33_BCLKON; + break; + } + dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); - reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); - if (dac33->fifo_mode) - reg_tmp &= ~DAC33_BCLKON; - else - reg_tmp |= DAC33_BCLKON; - dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); - if (dac33->fifo_mode) { + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: /* 20: BCLK divide ratio */ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); dac33_write16(codec, DAC33_ATHR_MSB, DAC33_THRREG(dac33->alarm_threshold)); - } else { + break; + default: + /* BYPASS mode */ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + break; } mutex_unlock(&dac33->mutex); -- cgit v0.10.2 From 28e05d987028023b09652bfe3ac597de6dba5e60 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:22 +0200 Subject: ASoC: tlv320dac33: Add new FIFO mode: mode 7 Mode 7 of tlv320dac33 operates in the following way: The codec is in master mode. Host configures upper and lower thresholds in tlv320dac33 During playback the codec will clock in the data until the upper threshold is reached in FIFO. At this point the codec stops the colocks on the serial bus. When the FIFO fill is reaching the lower threshold limit the codec will enable the clocks on the serial bus, and clocks in data till the upper threshold is reached. In this mode, we can also request interrupts for threshold events (upper, lower and alarm), which could be used for power management. At this point the interrupts are not enabled for this mode, but it can be taken into use in the future, when the surrounding code makes it possible to use it. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c684aa2..bc35f3f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -62,6 +62,7 @@ enum dac33_state { enum dac33_fifo_modes { DAC33_FIFO_BYPASS = 0, DAC33_FIFO_MODE1, + DAC33_FIFO_MODE7, DAC33_FIFO_LAST_MODE, }; @@ -422,7 +423,7 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol, /* Codec operation modes */ static const char *dac33_fifo_mode_texts[] = { - "Bypass", "Mode 1" + "Bypass", "Mode 1", "Mode 7" }; static const struct soc_enum dac33_fifo_mode_enum = @@ -556,6 +557,10 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(dac33->alarm_threshold)); break; + case DAC33_FIFO_MODE7: + dac33_write16(codec, DAC33_PREFILL_MSB, + DAC33_THRREG(20)); + break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", dac33->fifo_mode); @@ -574,6 +579,9 @@ static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) dac33_write16(codec, DAC33_NSAMPLE_MSB, DAC33_THRREG(dac33->nsample)); break; + case DAC33_FIFO_MODE7: + /* At the moment we are not using interrupts in mode7 */ + break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", dac33->fifo_mode); @@ -788,6 +796,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); break; + case DAC33_FIFO_MODE7: + /* Disable all interrupts */ + dac33_write(codec, DAC33_FIFO_IRQ_MASK, 0); + break; default: /* in FIFO bypass mode, the interrupts are not used */ break; @@ -807,6 +819,17 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) fifoctrl_a &= ~DAC33_FAUTO; aictrl_b &= ~DAC33_BCLKON; break; + case DAC33_FIFO_MODE7: + /* + * For mode1: + * Disable the FIFO bypass (Enable the use of FIFO) + * Select Threshold mode + * BCLK is only running when data is needed by DAC33 + */ + fifoctrl_a &= ~DAC33_FBYPAS; + fifoctrl_a |= DAC33_FAUTO; + aictrl_b &= ~DAC33_BCLKON; + break; default: /* * For FIFO bypass mode: @@ -830,6 +853,16 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write16(codec, DAC33_ATHR_MSB, DAC33_THRREG(dac33->alarm_threshold)); break; + case DAC33_FIFO_MODE7: + /* + * Configure the threshold levels, and leave 10 sample space + * at the bottom, and also at the top of the FIFO + */ + dac33_write16(codec, DAC33_UTHR_MSB, + DAC33_THRREG(DAC33_BUFFER_SIZE_SAMPLES - 10)); + dac33_write16(codec, DAC33_LTHR_MSB, + DAC33_THRREG(10)); + break; default: /* BYPASS mode */ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); -- cgit v0.10.2 From adcb8bc02d86259c117a03b54e9918e5ad3121af Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Dec 2009 10:30:23 +0200 Subject: ASoC: tlv320dac33: Safety check for codec slave mode The currently available FIFO modes (mode1 and mode7) require master mode from the codec. Do not allow the slave configuration when the FIFO is in use. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index bc35f3f..3ef3255 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -993,6 +993,7 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; u8 aictrl_a, aictrl_b; aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); @@ -1005,7 +1006,11 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, break; case SND_SOC_DAIFMT_CBS_CFS: /* Codec Slave */ - aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); + if (dac33->fifo_mode) { + dev_err(codec->dev, "FIFO mode requires master mode\n"); + return -EINVAL; + } else + aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); break; default: return -EINVAL; -- cgit v0.10.2 From 633154d3a7bbd542465b905392bf76b780f00b4f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Dec 2009 13:42:43 +0000 Subject: ASoC: Remove unneeded suspend checks from CODEC drivers Better integration of the core with the device model means that we now no longer get the ASoC suspend and resume callbacks without the card having been set up. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d6850da..c2444e7 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1507,10 +1507,6 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - /* we only need to suspend if we are a valid card */ - if (!codec->card) - return 0; - wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1523,10 +1519,6 @@ static int wm8753_resume(struct platform_device *pdev) u8 data[2]; u16 *cache = codec->reg_cache; - /* we only need to resume if we are a valid card */ - if (!codec->card) - return 0; - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) { if (i + 1 == WM8753_RESET) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 341481e..a54dc77b 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1319,10 +1319,6 @@ static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - /* we only need to suspend if we are a valid card */ - if (!codec->card) - return 0; - wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1335,10 +1331,6 @@ static int wm8990_resume(struct platform_device *pdev) u8 data[2]; u16 *cache = codec->reg_cache; - /* we only need to resume if we are a valid card */ - if (!codec->card) - return 0; - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { if (i + 1 == WM8990_RESET) -- cgit v0.10.2 From 40ca114265a281d51b261771df551a373fc8ff3c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Dec 2009 13:44:28 +0000 Subject: ASoC: Use snprintf() when generating stream names Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8b900a84..9b36c5e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1276,8 +1276,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, codec_dai->codec = card->codec; /* check client and interface hw capabilities */ - sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name, - num); + snprintf(new_name, sizeof(new_name), "%s %s-%d", + dai_link->stream_name, codec_dai->name, num); if (codec_dai->playback.channels_min) playback = 1; -- cgit v0.10.2 From a126fd5691e6cd680758b72e6ea288bb83b9deb6 Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Mon, 4 Jan 2010 14:30:03 +0200 Subject: ASoc: tpa6130a2: Remove unnecessary variable Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0eb33d4..8e98ccf 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -267,12 +267,8 @@ static const struct snd_kcontrol_new tpa6130a2_controls[] = { */ static void tpa6130a2_channel_enable(u8 channel, int enable) { - struct tpa6130a2_data *data; u8 val; - BUG_ON(tpa6130a2_client == NULL); - data = i2c_get_clientdata(tpa6130a2_client); - if (enable) { /* Enable channel */ /* Enable amplifier */ -- cgit v0.10.2 From 5baf831541c61546c00e8d6f294cb10ed5d25e7d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:13:42 +0000 Subject: ASoC: Fix variable shadowing warning in TLV320AIC3x Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5a8f53c..e4b946a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -849,20 +849,20 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, * The term had to be converted to get * rid of the division by 10000; d = 0 here */ - int clk = (1000 * j * r) / p; + int tmp_clk = (1000 * j * r) / p; /* Check whether this values get closer than * the best ones we had before */ - if (abs(codec_clk - clk) < + if (abs(codec_clk - tmp_clk) < abs(codec_clk - last_clk)) { pll_j = j; pll_d = 0; pll_r = r; pll_p = p; - last_clk = clk; + last_clk = tmp_clk; } /* Early exit for exact matches */ - if (clk == codec_clk) + if (tmp_clk == codec_clk) goto found; } } -- cgit v0.10.2 From d11c5ab186310389b8e573be00279bab0a565d30 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:14:07 +0000 Subject: ASoC: Only restore non-default registers for WM8731 Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 3a49781..5a2619d 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -456,6 +456,9 @@ static int wm8731_resume(struct platform_device *pdev) /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { + if (cache[i] == wm8731_reg[i]) + continue; + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); -- cgit v0.10.2 From e0fb28e079b50f891b6c9db1c2bb25fef3268cf4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:14:23 +0000 Subject: ASoC: Only restore non-default registers for WM8776 Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ab2c0da..44e7d9d 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -406,6 +406,8 @@ static int wm8776_resume(struct platform_device *pdev) /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) { + if (cache[i] == wm8776_reg[i]) + continue; data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); -- cgit v0.10.2 From 10505634bfa74871118a21eef8617acad00e4019 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:14:45 +0000 Subject: ASoC: Only restore non-default registers for WM8961 Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index a8007d5..d2342c5 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1022,6 +1022,9 @@ static int wm8961_resume(struct platform_device *pdev) int i; for (i = 0; i < codec->reg_cache_size; i++) { + if (reg_cache[i] == wm8961_reg_defaults[i]) + continue; + if (i == WM8961_SOFTWARE_RESET) continue; -- cgit v0.10.2 From 53242c68333570631a15a69842851b458eca3d99 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Jan 2010 13:15:56 +0000 Subject: ASoC: Implement suspend and resume for WM8993 Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 5e32f2e..cd2bc05 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -227,6 +227,7 @@ struct wm8993_priv { int class_w_users; unsigned int fll_fref; unsigned int fll_fout; + int fll_src; }; static unsigned int wm8993_read_hw(struct snd_soc_codec *codec, u8 reg) @@ -506,6 +507,7 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, wm8993->fll_fref = Fref; wm8993->fll_fout = Fout; + wm8993->fll_src = source; return 0; } @@ -1480,9 +1482,74 @@ static int wm8993_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int wm8993_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8993_priv *wm8993 = codec->private_data; + int fll_fout = wm8993->fll_fout; + int fll_fref = wm8993->fll_fref; + int ret; + + /* Stop the FLL in an orderly fashion */ + ret = wm8993_set_fll(codec->dai, 0, 0, 0, 0); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to stop FLL\n"); + return ret; + } + + wm8993->fll_fout = fll_fout; + wm8993->fll_fref = fll_fref; + + wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8993_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8993_priv *wm8993 = codec->private_data; + u16 *cache = wm8993->reg_cache; + int i, ret; + + /* Restore the register settings */ + for (i = 1; i < WM8993_MAX_REGISTER; i++) { + if (cache[i] == wm8993_reg_defaults[i]) + continue; + snd_soc_write(codec, i, cache[i]); + } + + wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Restart the FLL? */ + if (wm8993->fll_fout) { + int fll_fout = wm8993->fll_fout; + int fll_fref = wm8993->fll_fref; + + wm8993->fll_fref = 0; + wm8993->fll_fout = 0; + + ret = wm8993_set_fll(codec->dai, 0, wm8993->fll_src, + fll_fref, fll_fout); + if (ret != 0) + dev_err(codec->dev, "Failed to restart FLL\n"); + } + + return 0; +} +#else +#define wm8993_suspend NULL +#define wm8993_resume NULL +#endif + struct snd_soc_codec_device soc_codec_dev_wm8993 = { .probe = wm8993_probe, .remove = wm8993_remove, + .suspend = wm8993_suspend, + .resume = wm8993_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8993); -- cgit v0.10.2 From 741b20cfb9109760937f403d18d731bfde31f56f Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 17 Dec 2009 17:34:39 +0100 Subject: ALSA: pcm_lib.c - convert second xrun_debug() parameter to use defines To increase code readability, convert send xrun_debug() argument to use defines. Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 30f4108..9621236 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -126,15 +126,22 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +#define XRUN_DEBUG_BASIC (1<<0) +#define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ +#define XRUN_DEBUG_JIFFIESCHECK (1<<2) /* do jiffies check */ +#define XRUN_DEBUG_PERIODUPDATE (1<<3) /* full period update info */ +#define XRUN_DEBUG_HWPTRUPDATE (1<<4) /* full hwptr update info */ + #ifdef CONFIG_SND_PCM_XRUN_DEBUG -#define xrun_debug(substream, mask) ((substream)->pstr->xrun_debug & (mask)) +#define xrun_debug(substream, mask) \ + ((substream)->pstr->xrun_debug & (mask)) #else #define xrun_debug(substream, mask) 0 #endif -#define dump_stack_on_xrun(substream) do { \ - if (xrun_debug(substream, 2)) \ - dump_stack(); \ +#define dump_stack_on_xrun(substream) do { \ + if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ + dump_stack(); \ } while (0) static void pcm_debug_name(struct snd_pcm_substream *substream, @@ -154,7 +161,7 @@ static void xrun(struct snd_pcm_substream *substream) if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - if (xrun_debug(substream, 1)) { + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd(KERN_DEBUG "XRUN: %s\n", name); @@ -215,7 +222,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, #define hw_ptr_error(substream, fmt, args...) \ do { \ - if (xrun_debug(substream, 1)) { \ + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ if (printk_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ @@ -237,7 +244,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - if (xrun_debug(substream, 8)) { + if (xrun_debug(substream, XRUN_DEBUG_PERIODUPDATE)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " @@ -290,7 +297,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) } /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, 4)) + if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) goto no_jiffies_check; /* Skip the jiffies check for hardwares with BATCH flag. @@ -369,7 +376,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - if (xrun_debug(substream, 16)) { + if (xrun_debug(substream, XRUN_DEBUG_HWPTRUPDATE)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " @@ -403,7 +410,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, 4)) + if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) goto no_jiffies_check; if (delta < runtime->delay) goto no_jiffies_check; -- cgit v0.10.2 From 4d96eb255c53ab5e39b37fd4d484ea3dc39ab456 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 20 Dec 2009 11:47:57 +0100 Subject: ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions In some debug cases, it might be usefull to see previous ring buffer positions to determine position problems from the lowlevel drivers. Signed-off-by: Jaroslav Kysela diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a7..4e18a6d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -262,6 +262,8 @@ struct snd_pcm_hw_constraint_list { unsigned int mask; }; +struct snd_pcm_hwptr_log; + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -340,6 +342,10 @@ struct snd_pcm_runtime { /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif + +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + struct snd_pcm_hwptr_log *hwptr_log; +#endif }; struct snd_pcm_group { /* keep linked substreams */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6884ae0..df57a0e 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -921,6 +921,10 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) snd_free_pages((void*)runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + if (runtime->hwptr_log) + kfree(runtime->hwptr_log); +#endif kfree(runtime); substream->runtime = NULL; put_pid(substream->pid); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9621236..1990afb 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -126,34 +126,34 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +static void pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len) +{ + snprintf(name, len, "pcmC%dD%d%c:%d", + substream->pcm->card->number, + substream->pcm->device, + substream->stream ? 'c' : 'p', + substream->number); +} + #define XRUN_DEBUG_BASIC (1<<0) #define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ #define XRUN_DEBUG_JIFFIESCHECK (1<<2) /* do jiffies check */ #define XRUN_DEBUG_PERIODUPDATE (1<<3) /* full period update info */ #define XRUN_DEBUG_HWPTRUPDATE (1<<4) /* full hwptr update info */ +#define XRUN_DEBUG_LOG (1<<5) /* show last 10 positions on err */ +#define XRUN_DEBUG_LOGONCE (1<<6) /* do above only once */ #ifdef CONFIG_SND_PCM_XRUN_DEBUG + #define xrun_debug(substream, mask) \ ((substream)->pstr->xrun_debug & (mask)) -#else -#define xrun_debug(substream, mask) 0 -#endif #define dump_stack_on_xrun(substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ dump_stack(); \ } while (0) -static void pcm_debug_name(struct snd_pcm_substream *substream, - char *name, size_t len) -{ - snprintf(name, len, "pcmC%dD%d%c:%d", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p', - substream->number); -} - static void xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -169,6 +169,102 @@ static void xrun(struct snd_pcm_substream *substream) } } +#define hw_ptr_error(substream, fmt, args...) \ + do { \ + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ + if (printk_ratelimit()) { \ + snd_printd("PCM: " fmt, ##args); \ + } \ + dump_stack_on_xrun(substream); \ + } \ + } while (0) + +#define XRUN_LOG_CNT 10 + +struct hwptr_log_entry { + unsigned long jiffies; + snd_pcm_uframes_t pos; + snd_pcm_uframes_t period_size; + snd_pcm_uframes_t buffer_size; + snd_pcm_uframes_t old_hw_ptr; + snd_pcm_uframes_t hw_ptr_base; + snd_pcm_uframes_t hw_ptr_interrupt; +}; + +struct snd_pcm_hwptr_log { + unsigned int idx; + unsigned int hit: 1; + struct hwptr_log_entry entries[XRUN_LOG_CNT]; +}; + +static void xrun_log(struct snd_pcm_substream *substream, + snd_pcm_uframes_t pos) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hwptr_log *log = runtime->hwptr_log; + struct hwptr_log_entry *entry; + + if (log == NULL) { + log = kzalloc(sizeof(*log), GFP_ATOMIC); + if (log == NULL) + return; + runtime->hwptr_log = log; + } else { + if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) + return; + } + entry = &log->entries[log->idx]; + entry->jiffies = jiffies; + entry->pos = pos; + entry->period_size = runtime->period_size; + entry->buffer_size = runtime->buffer_size;; + entry->old_hw_ptr = runtime->status->hw_ptr; + entry->hw_ptr_base = runtime->hw_ptr_base; + entry->hw_ptr_interrupt = runtime->hw_ptr_interrupt;; + log->idx = (log->idx + 1) % XRUN_LOG_CNT; +} + +static void xrun_log_show(struct snd_pcm_substream *substream) +{ + struct snd_pcm_hwptr_log *log = substream->runtime->hwptr_log; + struct hwptr_log_entry *entry; + char name[16]; + unsigned int idx; + int cnt; + + if (log == NULL) + return; + if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) + return; + pcm_debug_name(substream, name, sizeof(name)); + for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { + entry = &log->entries[idx]; + if (entry->period_size == 0) + break; + snd_printd("hwptr log: %s: j=%lu, pos=0x%lx/0x%lx/0x%lx, " + "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + name, entry->jiffies, (unsigned long)entry->pos, + (unsigned long)entry->period_size, + (unsigned long)entry->buffer_size, + (unsigned long)entry->old_hw_ptr, + (unsigned long)entry->hw_ptr_base, + (unsigned long)entry->hw_ptr_interrupt); + idx++; + idx %= XRUN_LOG_CNT; + } + log->hit = 1; +} + +#else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ + +#define xrun_debug(substream, mask) 0 +#define xrun(substream) do { } while (0) +#define hw_ptr_error(substream, fmt, args...) do { } while (0) +#define xrun_log(substream, pos) do { } while (0) +#define xrun_log_show(substream) do { } while (0) + +#endif + static snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) @@ -182,6 +278,7 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, if (printk_ratelimit()) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); + xrun_log_show(substream); snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " "buffer size = 0x%lx, period size = 0x%lx\n", name, pos, runtime->buffer_size, @@ -190,6 +287,8 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, pos = 0; } pos -= pos % runtime->min_align; + if (xrun_debug(substream, XRUN_DEBUG_LOG)) + xrun_log(substream, pos); return pos; } @@ -220,16 +319,6 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return 0; } -#define hw_ptr_error(substream, fmt, args...) \ - do { \ - if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ - if (printk_ratelimit()) { \ - snd_printd("PCM: " fmt, ##args); \ - } \ - dump_stack_on_xrun(substream); \ - } \ - } while (0) - static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -270,6 +359,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) delta += runtime->buffer_size; if (delta < 0) { + xrun_log_show(substream); hw_ptr_error(substream, "Unexpected hw_pointer value " "(stream=%i, pos=%ld, intr_ptr=%ld)\n", @@ -315,6 +405,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) + HZ/100); + xrun_log_show(substream); hw_ptr_error(substream, "hw_ptr skipping! [Q] " "(pos=%ld, delta=%ld, period=%ld, " @@ -334,6 +425,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { + xrun_log_show(substream); hw_ptr_error(substream, "Lost interrupts? " "(stream=%i, delta=%ld, intr_ptr=%ld)\n", @@ -397,6 +489,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) if (delta < 0) { delta += runtime->buffer_size; if (delta < 0) { + xrun_log_show(substream); hw_ptr_error(substream, "Unexpected hw_pointer value [2] " "(stream=%i, pos=%ld, old_ptr=%ld, jdelta=%li)\n", @@ -416,6 +509,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) goto no_jiffies_check; delta -= runtime->delay; if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { + xrun_log_show(substream); hw_ptr_error(substream, "hw_ptr skipping! " "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", -- cgit v0.10.2 From f240406babfe1526998e10583ea5eccc2676a433 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 5 Jan 2010 17:19:34 +0100 Subject: ALSA: pcm_lib - cleanup & merge hw_ptr update functions Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them. The main change is hw_ptr_interrupt variable removal to simplify code logic. This variable can be computed directly from hw_ptr. Ensure that updated hw_ptr is not lower than previous one (it was possible with old code in some obscure situations when interrupt was delayed or the lowlevel driver returns wrong ring buffer position value). Signed-off-by: Jaroslav Kysela diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 4e18a6d..fe1b131 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -271,7 +271,6 @@ struct snd_pcm_runtime { int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ - snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index cc4e226..760c969 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -61,7 +61,7 @@ struct snd_pcm_oss_runtime { struct snd_pcm_plugin *plugin_first; struct snd_pcm_plugin *plugin_last; #endif - unsigned int prev_hw_ptr_interrupt; + unsigned int prev_hw_ptr_period; }; struct snd_pcm_oss_file { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d9c9635..255ad91 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -632,6 +632,13 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes); } +static inline +snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) +{ + snd_pcm_uframes_t ptr = runtime->status->hw_ptr; + return ptr - (ptr % runtime->period_size); +} + /* define extended formats in the recent OSS versions (if any) */ /* linear formats */ #define AFMT_S32_LE 0x00001000 @@ -1102,7 +1109,7 @@ static int snd_pcm_oss_prepare(struct snd_pcm_substream *substream) return err; } runtime->oss.prepare = 0; - runtime->oss.prev_hw_ptr_interrupt = 0; + runtime->oss.prev_hw_ptr_period = 0; runtime->oss.period_ptr = 0; runtime->oss.buffer_used = 0; @@ -1950,7 +1957,8 @@ static int snd_pcm_oss_get_caps(struct snd_pcm_oss_file *pcm_oss_file) return result; } -static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, snd_pcm_uframes_t hw_ptr) +static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, + snd_pcm_uframes_t hw_ptr) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t appl_ptr; @@ -1986,7 +1994,8 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr if (runtime->oss.trigger) goto _skip1; if (atomic_read(&psubstream->mmap_count)) - snd_pcm_oss_simulate_fill(psubstream, runtime->hw_ptr_interrupt); + snd_pcm_oss_simulate_fill(psubstream, + get_hw_ptr_period(runtime)); runtime->oss.trigger = 1; runtime->start_threshold = 1; cmd = SNDRV_PCM_IOCTL_START; @@ -2105,11 +2114,12 @@ static int snd_pcm_oss_get_ptr(struct snd_pcm_oss_file *pcm_oss_file, int stream info.ptr = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr % runtime->buffer_size); if (atomic_read(&substream->mmap_count)) { snd_pcm_sframes_t n; - n = (delay = runtime->hw_ptr_interrupt) - runtime->oss.prev_hw_ptr_interrupt; + delay = get_hw_ptr_period(runtime); + n = delay - runtime->oss.prev_hw_ptr_period; if (n < 0) n += runtime->boundary; info.blocks = n / runtime->period_size; - runtime->oss.prev_hw_ptr_interrupt = delay; + runtime->oss.prev_hw_ptr_period = delay; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_pcm_oss_simulate_fill(substream, delay); info.bytes = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr) & INT_MAX; @@ -2673,18 +2683,22 @@ static int snd_pcm_oss_playback_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) - return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; + return runtime->oss.prev_hw_ptr_period != + get_hw_ptr_period(runtime); else - return snd_pcm_playback_avail(runtime) >= runtime->oss.period_frames; + return snd_pcm_playback_avail(runtime) >= + runtime->oss.period_frames; } static int snd_pcm_oss_capture_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) - return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; + return runtime->oss.prev_hw_ptr_period != + get_hw_ptr_period(runtime); else - return snd_pcm_capture_avail(runtime) >= runtime->oss.period_frames; + return snd_pcm_capture_avail(runtime) >= + runtime->oss.period_frames; } static unsigned int snd_pcm_oss_poll(struct file *file, poll_table * wait) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1990afb..70a4f74 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -172,6 +172,7 @@ static void xrun(struct snd_pcm_substream *substream) #define hw_ptr_error(substream, fmt, args...) \ do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ + xrun_log_show(substream); \ if (printk_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ @@ -188,7 +189,6 @@ struct hwptr_log_entry { snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t old_hw_ptr; snd_pcm_uframes_t hw_ptr_base; - snd_pcm_uframes_t hw_ptr_interrupt; }; struct snd_pcm_hwptr_log { @@ -220,7 +220,6 @@ static void xrun_log(struct snd_pcm_substream *substream, entry->buffer_size = runtime->buffer_size;; entry->old_hw_ptr = runtime->status->hw_ptr; entry->hw_ptr_base = runtime->hw_ptr_base; - entry->hw_ptr_interrupt = runtime->hw_ptr_interrupt;; log->idx = (log->idx + 1) % XRUN_LOG_CNT; } @@ -241,14 +240,13 @@ static void xrun_log_show(struct snd_pcm_substream *substream) entry = &log->entries[idx]; if (entry->period_size == 0) break; - snd_printd("hwptr log: %s: j=%lu, pos=0x%lx/0x%lx/0x%lx, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + snd_printd("hwptr log: %s: j=%lu, pos=%ld/%ld/%ld, " + "hwptr=%ld/%ld\n", name, entry->jiffies, (unsigned long)entry->pos, (unsigned long)entry->period_size, (unsigned long)entry->buffer_size, (unsigned long)entry->old_hw_ptr, - (unsigned long)entry->hw_ptr_base, - (unsigned long)entry->hw_ptr_interrupt); + (unsigned long)entry->hw_ptr_base); idx++; idx %= XRUN_LOG_CNT; } @@ -265,33 +263,6 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #endif -static snd_pcm_uframes_t -snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) -{ - snd_pcm_uframes_t pos; - - pos = substream->ops->pointer(substream); - if (pos == SNDRV_PCM_POS_XRUN) - return pos; /* XRUN */ - if (pos >= runtime->buffer_size) { - if (printk_ratelimit()) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - xrun_log_show(substream); - snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " - "buffer size = 0x%lx, period size = 0x%lx\n", - name, pos, runtime->buffer_size, - runtime->period_size); - } - pos = 0; - } - pos -= pos % runtime->min_align; - if (xrun_debug(substream, XRUN_DEBUG_LOG)) - xrun_log(substream, pos); - return pos; -} - static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) { @@ -319,72 +290,88 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return 0; } -static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) +static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, + unsigned int in_interrupt) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_ptr_interrupt, hw_base; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t hdelta, delta; unsigned long jdelta; old_hw_ptr = runtime->status->hw_ptr; - pos = snd_pcm_update_hw_ptr_pos(substream, runtime); + pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); return -EPIPE; } - if (xrun_debug(substream, XRUN_DEBUG_PERIODUPDATE)) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", - name, (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)old_hw_ptr, - (unsigned long)runtime->hw_ptr_base, - (unsigned long)runtime->hw_ptr_interrupt); + if (pos >= runtime->buffer_size) { + if (printk_ratelimit()) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + xrun_log_show(substream); + snd_printd(KERN_ERR "BUG: %s, pos = %ld, " + "buffer size = %ld, period size = %ld\n", + name, pos, runtime->buffer_size, + runtime->period_size); + } + pos = 0; } + pos -= pos % runtime->min_align; + if (xrun_debug(substream, XRUN_DEBUG_LOG)) + xrun_log(substream, pos); hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; - hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - delta = new_hw_ptr - hw_ptr_interrupt; - if (hw_ptr_interrupt >= runtime->boundary) { - hw_ptr_interrupt -= runtime->boundary; - if (hw_base < runtime->boundary / 2) - /* hw_base was already lapped; recalc delta */ - delta = new_hw_ptr - hw_ptr_interrupt; - } - if (delta < 0) { - if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) - delta += runtime->buffer_size; - if (delta < 0) { - xrun_log_show(substream); - hw_ptr_error(substream, - "Unexpected hw_pointer value " - "(stream=%i, pos=%ld, intr_ptr=%ld)\n", - substream->stream, (long)pos, - (long)hw_ptr_interrupt); -#if 1 - /* simply skipping the hwptr update seems more - * robust in some cases, e.g. on VMware with - * inaccurate timer source - */ - return 0; /* skip this update */ -#else - /* rebase to interrupt position */ - hw_base = new_hw_ptr = hw_ptr_interrupt; - /* align hw_base to buffer_size */ - hw_base -= hw_base % runtime->buffer_size; - delta = 0; -#endif - } else { + if (in_interrupt) { + /* we know that one period was processed */ + /* delta = "expected next hw_ptr" for in_interrupt != 0 */ + delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) + + runtime->period_size; + if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; + goto __delta; } } + /* new_hw_ptr might be lower than old_hw_ptr in case when */ + /* pointer crosses the end of the ring buffer */ + if (new_hw_ptr < old_hw_ptr) { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + __delta: + delta = (new_hw_ptr - old_hw_ptr) % runtime->boundary; + if (xrun_debug(substream, in_interrupt ? + XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd("%s_update: %s: pos=%u/%u/%u, " + "hwptr=%ld/%ld/%ld/%ld\n", + in_interrupt ? "period" : "hwptr", + name, + (unsigned int)pos, + (unsigned int)runtime->period_size, + (unsigned int)runtime->buffer_size, + (unsigned long)delta, + (unsigned long)old_hw_ptr, + (unsigned long)new_hw_ptr, + (unsigned long)runtime->hw_ptr_base); + } + /* something must be really wrong */ + if (delta >= runtime->buffer_size) { + hw_ptr_error(substream, + "Unexpected hw_pointer value %s" + "(stream=%i, pos=%ld, new_hw_ptr=%ld, " + "old_hw_ptr=%ld)\n", + in_interrupt ? "[Q] " : "[P]", + substream->stream, (long)pos, + (long)new_hw_ptr, (long)old_hw_ptr); + return 0; + } /* Do jiffies check only in xrun_debug mode */ if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) @@ -396,7 +383,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) */ if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; - hdelta = new_hw_ptr - old_hw_ptr; + hdelta = delta; if (hdelta < runtime->delay) goto no_jiffies_check; hdelta -= runtime->delay; @@ -405,45 +392,49 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) + HZ/100); - xrun_log_show(substream); + /* move new_hw_ptr according jiffies not pos variable */ + new_hw_ptr = old_hw_ptr; + /* use loop to avoid checks for delta overflows */ + /* the delta value is small or zero in most cases */ + while (delta > 0) { + new_hw_ptr += runtime->period_size; + if (new_hw_ptr >= runtime->boundary) + new_hw_ptr -= runtime->boundary; + delta--; + } + /* align hw_base to buffer_size */ + hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); + delta = 0; hw_ptr_error(substream, - "hw_ptr skipping! [Q] " + "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " - "jdelta=%lu/%lu/%lu)\n", + "jdelta=%lu/%lu/%lu, hw_ptr=%ld/%ld)\n", + in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, - ((hdelta * HZ) / runtime->rate), delta); - hw_ptr_interrupt = runtime->hw_ptr_interrupt + - runtime->period_size * delta; - if (hw_ptr_interrupt >= runtime->boundary) - hw_ptr_interrupt -= runtime->boundary; - /* rebase to interrupt position */ - hw_base = new_hw_ptr = hw_ptr_interrupt; - /* align hw_base to buffer_size */ - hw_base -= hw_base % runtime->buffer_size; - delta = 0; + ((hdelta * HZ) / runtime->rate), delta, + (unsigned long)old_hw_ptr, + (unsigned long)new_hw_ptr); } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { - xrun_log_show(substream); hw_ptr_error(substream, - "Lost interrupts? " - "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + "Lost interrupts? %s" + "(stream=%i, delta=%ld, new_hw_ptr=%ld, " + "old_hw_ptr=%ld)\n", + in_interrupt ? "[Q] " : "", substream->stream, (long)delta, - (long)hw_ptr_interrupt); - /* rebase hw_ptr_interrupt */ - hw_ptr_interrupt = - new_hw_ptr - new_hw_ptr % runtime->period_size; + (long)new_hw_ptr, + (long)old_hw_ptr); } - runtime->hw_ptr_interrupt = hw_ptr_interrupt; + + if (runtime->status->hw_ptr == new_hw_ptr) + return 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); - if (runtime->status->hw_ptr == new_hw_ptr) - return 0; - runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; @@ -456,83 +447,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) /* CAUTION: call it with irq disabled */ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; - snd_pcm_sframes_t delta; - unsigned long jdelta; - - old_hw_ptr = runtime->status->hw_ptr; - pos = snd_pcm_update_hw_ptr_pos(substream, runtime); - if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); - return -EPIPE; - } - if (xrun_debug(substream, XRUN_DEBUG_HWPTRUPDATE)) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", - name, (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)old_hw_ptr, - (unsigned long)runtime->hw_ptr_base, - (unsigned long)runtime->hw_ptr_interrupt); - } - - hw_base = runtime->hw_ptr_base; - new_hw_ptr = hw_base + pos; - - delta = new_hw_ptr - old_hw_ptr; - jdelta = jiffies - runtime->hw_ptr_jiffies; - if (delta < 0) { - delta += runtime->buffer_size; - if (delta < 0) { - xrun_log_show(substream); - hw_ptr_error(substream, - "Unexpected hw_pointer value [2] " - "(stream=%i, pos=%ld, old_ptr=%ld, jdelta=%li)\n", - substream->stream, (long)pos, - (long)old_hw_ptr, jdelta); - return 0; - } - hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) - hw_base = 0; - new_hw_ptr = hw_base + pos; - } - /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) - goto no_jiffies_check; - if (delta < runtime->delay) - goto no_jiffies_check; - delta -= runtime->delay; - if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { - xrun_log_show(substream); - hw_ptr_error(substream, - "hw_ptr skipping! " - "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", - (long)pos, (long)delta, - (long)runtime->period_size, jdelta, - ((delta * HZ) / runtime->rate)); - return 0; - } - no_jiffies_check: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream, new_hw_ptr); - - if (runtime->status->hw_ptr == new_hw_ptr) - return 0; - - runtime->hw_ptr_base = hw_base; - runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_jiffies = jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) - snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); - - return snd_pcm_update_hw_ptr_post(substream, runtime); + return snd_pcm_update_hw_ptr0(substream, 0); } /** @@ -1744,7 +1659,7 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || - snd_pcm_update_hw_ptr_interrupt(substream) < 0) + snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; if (substream->timer_running) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 29ab46a1..8e777f7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1247,8 +1247,6 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) if (err < 0) return err; runtime->hw_ptr_base = 0; - runtime->hw_ptr_interrupt = runtime->status->hw_ptr - - runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; return 0; -- cgit v0.10.2 From 1250932e48d3b698415b1f04775433cf1da688d6 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 7 Jan 2010 15:36:31 +0100 Subject: ALSA: pcm_lib - optimize wake_up() calls for PCM I/O As noted by pl bossart , the PCM I/O routines (snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls until all samples are not processed. Signed-off-by: Jaroslav Kysela diff --git a/include/sound/pcm.h b/include/sound/pcm.h index fe1b131..e26fb3c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -311,6 +311,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ + unsigned int nowake: 1; /* no wakeup (data-copy in progress) */ wait_queue_head_t sleep; struct fasync_struct *fasync; @@ -839,6 +840,8 @@ void snd_pcm_set_sync(struct snd_pcm_substream *substream); int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg); +int snd_pcm_update_state(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); int snd_pcm_playback_xrun_check(struct snd_pcm_substream *substream); int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 70a4f74..a632262 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -263,8 +263,8 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #endif -static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +int snd_pcm_update_state(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t avail; @@ -285,7 +285,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) + if (!runtime->nowake && avail >= runtime->control->avail_min) wake_up(&runtime->sleep); return 0; } @@ -441,7 +441,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); - return snd_pcm_update_hw_ptr_post(substream, runtime); + return snd_pcm_update_state(substream, runtime); } /* CAUTION: call it with irq disabled */ @@ -1792,6 +1792,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } + runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1813,15 +1814,17 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { + runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } appl_ptr = runtime->control->appl_ptr; appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); - if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0) - goto _end; + err = transfer(substream, appl_ofs, data, offset, frames); snd_pcm_stream_lock_irq(substream); + if (err < 0) + goto _end_unlock; switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE; @@ -1850,8 +1853,10 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } } _end_unlock: + runtime->nowake = 0; + if (xfer > 0 && err >= 0) + snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - _end: return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } @@ -2009,6 +2014,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } + runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2037,15 +2043,17 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { + runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } appl_ptr = runtime->control->appl_ptr; appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); - if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0) - goto _end; + err = transfer(substream, appl_ofs, data, offset, frames); snd_pcm_stream_lock_irq(substream); + if (err < 0) + goto _end_unlock; switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE; @@ -2068,8 +2076,10 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, xfer += frames; } _end_unlock: + runtime->nowake = 0; + if (xfer > 0 && err >= 0) + snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - _end: return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 8e777f7..27284f6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -516,6 +516,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, struct snd_pcm_sw_params *params) { struct snd_pcm_runtime *runtime; + int err; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; @@ -540,6 +541,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (params->silence_threshold > runtime->buffer_size) return -EINVAL; } + err = 0; snd_pcm_stream_lock_irq(substream); runtime->tstamp_mode = params->tstamp_mode; runtime->period_step = params->period_step; @@ -553,10 +555,10 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - wake_up(&runtime->sleep); + err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); - return 0; + return err; } static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream, -- cgit v0.10.2 From 7b3a177b0d4f92b3431b8dca777313a07533a710 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 8 Jan 2010 08:43:01 +0100 Subject: ALSA: pcm_lib: fix "something must be really wrong" condition When runtime->periods == 1 or when pointer crosses end of ring buffer, the delta might be greater than buffer_size. Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a632262..c7b35b2 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -362,7 +362,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (unsigned long)runtime->hw_ptr_base); } /* something must be really wrong */ - if (delta >= runtime->buffer_size) { + if (delta >= runtime->buffer_size + runtime->period_size) { hw_ptr_error(substream, "Unexpected hw_pointer value %s" "(stream=%i, pos=%ld, new_hw_ptr=%ld, " -- cgit v0.10.2 From dd3533eca859a6debb1565503ec03e68354e08e0 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 1 Jan 2010 19:05:43 +0100 Subject: ALSA: ac97_codec: merge WM9703 and WM9705 ops The WM9705 and WM9703 ops are the same actually so use the same code for both. Signed-off-by: Krzysztof Helt Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 139cf3b..e288a55 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -544,25 +544,10 @@ static int patch_wolfson04(struct snd_ac97 * ac97) return 0; } -static int patch_wolfson_wm9705_specific(struct snd_ac97 * ac97) -{ - int err, i; - for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0) - return err; - } - snd_ac97_write_cache(ac97, 0x72, 0x0808); - return 0; -} - -static struct snd_ac97_build_ops patch_wolfson_wm9705_ops = { - .build_specific = patch_wolfson_wm9705_specific, -}; - static int patch_wolfson05(struct snd_ac97 * ac97) { /* WM9705, WM9710 */ - ac97->build_ops = &patch_wolfson_wm9705_ops; + ac97->build_ops = &patch_wolfson_wm9703_ops; #ifdef CONFIG_TOUCHSCREEN_WM9705 /* WM9705 touchscreen uses AUX and VIDEO for touch */ ac97->flags |= AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; -- cgit v0.10.2 From cd9d95a55550555da8e587ead9cbba5f98a371a3 Mon Sep 17 00:00:00 2001 From: Ken Prox Date: Fri, 8 Jan 2010 09:01:47 +0100 Subject: ALSA: hda - conexant - Fixed microphone mixer for HP Compaq Presario F700 Added patch for Hewlett-Packard Company Device Subsystem id - 103c:30ea. Signed-off-by: Ken Prox Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1ab2958..b20c640 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1720,6 +1720,22 @@ static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_f700_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1810,6 +1826,32 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_f700_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1829,6 +1871,7 @@ enum { CXT5051_HP, /* no docking */ CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ + CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_MODELS }; @@ -1837,6 +1880,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", + [CXT5051_F700] = "hp 700" }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1846,6 +1890,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), + SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), {} }; @@ -1896,6 +1941,11 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; break; + case CXT5051_F700: + spec->init_verbs[0] = cxt5051_f700_init_verbs; + spec->mixers[0] = cxt5051_f700_mixers; + spec->no_auto_mic = 1; + break; } return 0; -- cgit v0.10.2 From 75f8991d0e6969407d51501d5a0537f104075c99 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Thu, 7 Jan 2010 13:46:25 +0100 Subject: ALSA: hda - Configure XO-1.5 microphones at capture time The XO-1.5 has a microphone LED designed to indicate to the user when something is being recorded. This light is controlled by the microphone bias voltage and it is currently coming on all the time. This patch defers the microphone port configuration until when recording is actually taking place, fixing the behaviour of the LED. Signed-off-by: Daniel Drake Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 01e46ba..3521f33 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -111,8 +111,12 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; - unsigned char ext_mic_bias; unsigned int dell_vostro; + + unsigned int ext_mic_present; + unsigned int recording; + void (*capture_prepare)(struct hda_codec *codec); + void (*capture_cleanup)(struct hda_codec *codec); }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -185,6 +189,8 @@ static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; + if (spec->capture_prepare) + spec->capture_prepare(codec); snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], stream_tag, 0, format); return 0; @@ -196,6 +202,8 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct conexant_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); + if (spec->capture_cleanup) + spec->capture_cleanup(codec); return 0; } @@ -2016,53 +2024,53 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -/* toggle input of built-in and mic jack appropriately */ -static void cxt5066_automic(struct hda_codec *codec) +/* OLPC defers mic widget control until when capture is started because the + * microphone LED comes on as soon as these settings are put in place. if we + * did this before recording, it would give the false indication that recording + * is happening when it is not. */ +static void cxt5066_olpc_select_mic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - struct hda_verb ext_mic_present[] = { - /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, - - /* switch to external mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + if (!spec->recording) + return; - /* disable internal mic, port C */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static struct hda_verb ext_mic_absent[] = { - /* enable internal mic, port C */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* external mic, port B */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); - /* switch to internal mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + /* internal mic, port C */ + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->ext_mic_present ? 0 : PIN_VREF80); +} - /* disable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; +/* toggle input of built-in and mic jack appropriately */ +static void cxt5066_olpc_automic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; unsigned int present; - present = snd_hda_jack_detect(codec, 0x1a); - if (present) { + present = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) snd_printdd("CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { + else snd_printdd("CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } + + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0 : 1); + spec->ext_mic_present = !!present; + + cxt5066_olpc_select_mic(codec); } /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_vostro_automic(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; unsigned int present; struct hda_verb ext_mic_present[] = { /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* switch to external mic input */ {0x17, AC_VERB_SET_CONNECT_SEL, 0}, @@ -2113,7 +2121,7 @@ static void cxt5066_hp_automute(struct hda_codec *codec) } /* unsolicited event for jack sensing */ -static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { @@ -2121,7 +2129,7 @@ static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_automic(codec); + cxt5066_olpc_automic(codec); break; } } @@ -2197,6 +2205,31 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, return 1; } +static void cxt5066_olpc_capture_prepare(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + /* mark as recording and configure the microphone widget so that the + * recording LED comes on. */ + spec->recording = 1; + cxt5066_olpc_select_mic(codec); +} + +static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + const struct hda_verb disable_mics[] = { + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* disble internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {}, + }; + + snd_hda_sequence_write(codec, disable_mics); + spec->recording = 0; +} + static struct hda_input_mux cxt5066_capture_source = { .num_items = 4, .items = { @@ -2347,10 +2380,10 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port C: internal microphone */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port D: unused */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, @@ -2479,12 +2512,19 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_hp_automute(codec); if (spec->dell_vostro) cxt5066_vostro_automic(codec); - else - cxt5066_automic(codec); } return 0; } +static int cxt5066_olpc_init(struct hda_codec *codec) +{ + snd_printdd("CXT5066: init\n"); + conexant_init(codec); + cxt5066_hp_automute(codec); + cxt5066_olpc_automic(codec); + return 0; +} + enum { CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ @@ -2521,7 +2561,7 @@ static int patch_cxt5066(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = cxt5066_init; + codec->patch_ops.init = conexant_init; spec->dell_automute = 0; spec->multiout.max_channels = 2; @@ -2534,7 +2574,6 @@ static int patch_cxt5066(struct hda_codec *codec) spec->input_mux = &cxt5066_capture_source; spec->port_d_mode = PIN_HP; - spec->ext_mic_bias = PIN_VREF80; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5066_init_verbs; @@ -2561,20 +2600,26 @@ static int patch_cxt5066(struct hda_codec *codec) spec->dell_automute = 1; break; case CXT5066_OLPC_XO_1_5: - codec->patch_ops.unsol_event = cxt5066_unsol_event; + codec->patch_ops.init = cxt5066_olpc_init; + codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; - spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; /* input source automatically selected */ spec->input_mux = NULL; + + /* our capture hooks which allow us to turn on the microphone LED + * at the right time */ + spec->capture_prepare = cxt5066_olpc_capture_prepare; + spec->capture_cleanup = cxt5066_olpc_capture_cleanup; break; case CXT5066_DELL_VOSTO: + codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_vostro_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; -- cgit v0.10.2 From c4cfe66c4c2d5a91b3734ffb4e2bad0badd5c874 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Thu, 7 Jan 2010 13:47:04 +0100 Subject: ALSA: hda - support OLPC XO-1.5 DC input The XO's audio hardware is wired up to allow DC sensors (e.g. light sensors, thermistors, etc) to be plugged in through the microphone jack. Add sound mixer controls to allow this mode to be enabled and tweaked. Signed-off-by: Daniel Drake Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3521f33..685015a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -117,6 +117,16 @@ struct conexant_spec { unsigned int recording; void (*capture_prepare)(struct hda_codec *codec); void (*capture_cleanup)(struct hda_codec *codec); + + /* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors) + * through the microphone jack. + * When the user enables this through a mixer switch, both internal and + * external microphones are disabled. Gain is fixed at 0dB. In this mode, + * we also allow the bias to be configured through a separate mixer + * control. */ + unsigned int dc_enable; + unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */ + unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -2024,6 +2034,26 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } +static const struct hda_input_mux cxt5066_olpc_dc_bias = { + .num_items = 3, + .items = { + { "Off", PIN_IN }, + { "50%", PIN_VREF50 }, + { "80%", PIN_VREF80 }, + }, +}; + +static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + /* Even though port F is the DC input, the bias is controlled on port B. + * we also leave that port as an active input (but unselected) in DC mode + * just in case that is necessary to make the bias setting take effect. */ + return snd_hda_codec_write_cache(codec, 0x1a, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index); +} + /* OLPC defers mic widget control until when capture is started because the * microphone LED comes on as soon as these settings are put in place. if we * did this before recording, it would give the false indication that recording @@ -2034,6 +2064,27 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec) if (!spec->recording) return; + if (spec->dc_enable) { + /* in DC mode we ignore presence detection and just use the jack + * through our special DC port */ + const struct hda_verb enable_dc_mode[] = { + /* disble internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable DC capture, port F */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {}, + }; + + snd_hda_sequence_write(codec, enable_dc_mode); + /* port B input disabled (and bias set) through the following call */ + cxt5066_set_olpc_dc_bias(codec); + return; + } + + /* disable DC (port F) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + /* external mic, port B */ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); @@ -2049,6 +2100,9 @@ static void cxt5066_olpc_automic(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int present; + if (spec->dc_enable) /* don't do presence detection in DC mode */ + return; + present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; if (present) @@ -2123,13 +2177,16 @@ static void cxt5066_hp_automute(struct hda_codec *codec) /* unsolicited event for jack sensing */ static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { + struct conexant_spec *spec = codec->spec; snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_olpc_automic(codec); + /* ignore mic events in DC mode; we're always using the jack */ + if (!spec->dc_enable) + cxt5066_olpc_automic(codec); break; } } @@ -2159,6 +2216,15 @@ static const struct hda_input_mux cxt5066_analog_mic_boost = { }, }; +static int cxt5066_set_mic_boost(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + return snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + cxt5066_analog_mic_boost.items[spec->mic_boost].index); +} + static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2169,15 +2235,8 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int val; - hda_nid_t nid = kcontrol->private_value & 0xff; - int inout = (kcontrol->private_value & 0x100) ? - AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT; - - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, inout); - - ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; + struct conexant_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->mic_boost; return 0; } @@ -2185,23 +2244,101 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; unsigned int idx; - hda_nid_t nid = kcontrol->private_value & 0xff; - int inout = (kcontrol->private_value & 0x100) ? - AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + + spec->mic_boost = idx; + if (!spec->dc_enable) + cxt5066_set_mic_boost(codec); + return 1; +} + +static void cxt5066_enable_dc(struct hda_codec *codec) +{ + const struct hda_verb enable_dc_mode[] = { + /* disable gain */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* switch to DC input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 3}, + {} + }; + + /* configure as input source */ + snd_hda_sequence_write(codec, enable_dc_mode); + cxt5066_olpc_select_mic(codec); /* also sets configured bias */ +} + +static void cxt5066_disable_dc(struct hda_codec *codec) +{ + /* reconfigure input source */ + cxt5066_set_mic_boost(codec); + /* automic also selects the right mic if we're recording */ + cxt5066_olpc_automic(codec); +} + +static int cxt5066_olpc_dc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = spec->dc_enable; + return 0; +} - if (!imux->num_items) +static int cxt5066_olpc_dc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + int dc_enable = !!ucontrol->value.integer.value[0]; + + if (dc_enable == spec->dc_enable) return 0; + + spec->dc_enable = dc_enable; + if (dc_enable) + cxt5066_enable_dc(codec); + else + cxt5066_disable_dc(codec); + + return 1; +} + +static int cxt5066_olpc_dc_bias_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_hda_input_mux_info(&cxt5066_olpc_dc_bias, uinfo); +} + +static int cxt5066_olpc_dc_bias_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->dc_input_bias; + return 0; +} + +static int cxt5066_olpc_dc_bias_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; + unsigned int idx; + idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout | - imux->items[idx].index); - + spec->dc_input_bias = idx; + if (spec->dc_enable) + cxt5066_set_olpc_dc_bias(codec); return 1; } @@ -2223,6 +2360,9 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) /* disble internal mic, port C */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* disable DC capture, port F */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {}, }; @@ -2282,6 +2422,24 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { {} }; +static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Mode Enable Switch", + .info = snd_ctl_boolean_mono_info, + .get = cxt5066_olpc_dc_get, + .put = cxt5066_olpc_dc_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Input Bias Enum", + .info = cxt5066_olpc_dc_bias_enum_info, + .get = cxt5066_olpc_dc_bias_enum_get, + .put = cxt5066_olpc_dc_bias_enum_put, + }, + {} +}; + static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2294,11 +2452,10 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Ext Mic Boost Capture Enum", + .name = "Analog Mic Boost Capture Enum", .info = cxt5066_mic_boost_mux_enum_info, .get = cxt5066_mic_boost_mux_enum_get, .put = cxt5066_mic_boost_mux_enum_put, - .private_value = 0x17, }, HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), @@ -2392,7 +2549,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - /* Port F: unused */ + /* Port F: external DC input through microphone port */ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port G: internal speakers */ @@ -2513,15 +2670,22 @@ static int cxt5066_init(struct hda_codec *codec) if (spec->dell_vostro) cxt5066_vostro_automic(codec); } + cxt5066_set_mic_boost(codec); return 0; } static int cxt5066_olpc_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; snd_printdd("CXT5066: init\n"); conexant_init(codec); cxt5066_hp_automute(codec); - cxt5066_olpc_automic(codec); + if (!spec->dc_enable) { + cxt5066_set_mic_boost(codec); + cxt5066_olpc_automic(codec); + } else { + cxt5066_enable_dc(codec); + } return 0; } @@ -2604,8 +2768,10 @@ static int patch_cxt5066(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_olpc_dc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; @@ -2627,6 +2793,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; spec->port_d_mode = 0; spec->dell_vostro = 1; + spec->mic_boost = 3; /* default 30dB gain */ snd_hda_attach_beep_device(codec, 0x13); /* no S/PDIF out */ -- cgit v0.10.2 From 2138301e1687bd4f22aa2b4df4829b6ffdae19bc Mon Sep 17 00:00:00 2001 From: Ilkka Koskinen Date: Fri, 8 Jan 2010 17:48:31 +0200 Subject: ASoC: tpa6130a2: Support for tpa6140's regulators tpa6140a2 uses different names for the regulators. Signed-off-by: Ilkka Koskinen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h index e8c901e..e29fde6 100644 --- a/include/sound/tpa6130a2-plat.h +++ b/include/sound/tpa6130a2-plat.h @@ -23,7 +23,13 @@ #ifndef TPA6130A2_PLAT_H #define TPA6130A2_PLAT_H +enum tpa_model { + TPA6130A2, + TPA6140A2, +}; + struct tpa6130a2_platform_data { + enum tpa_model id; int power_gpio; }; diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8e98ccf..8b27281 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -41,6 +41,11 @@ static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { "Vdd", }; +static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { + "HPVdd", + "AVdd", +}; + /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; @@ -420,8 +425,21 @@ static int tpa6130a2_probe(struct i2c_client *client, gpio_direction_output(data->power_gpio, 0); } - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + switch (pdata->id) { + case TPA6130A2: + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + break; + case TPA6140A2: + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6140a2_supply_names[i];; + break; + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + } ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), data->supplies); -- cgit v0.10.2 From 6b98515a620592636d2f8e0d3e2942d1cb4847ec Mon Sep 17 00:00:00 2001 From: Alan Cox Date: Mon, 4 Jan 2010 16:22:59 +0000 Subject: sound_oss: remove use of old BKL ioctl path Signed-off-by: Alan Cox Signed-off-by: Takashi Iwai diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 61aaeda..6c3267b 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -328,11 +328,11 @@ static int sound_mixer_ioctl(int mixdev, unsigned int cmd, void __user *arg) return mixer_devs[mixdev]->ioctl(mixdev, cmd, arg); } -static int sound_ioctl(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) +static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { int len = 0, dtype; - int dev = iminor(inode); + int dev = iminor(file->f_dentry->d_inode); + long ret = -EINVAL; void __user *p = (void __user *)arg; if (_SIOC_DIR(cmd) != _SIOC_NONE && _SIOC_DIR(cmd) != 0) { @@ -353,6 +353,7 @@ static int sound_ioctl(struct inode *inode, struct file *file, if (cmd == OSS_GETVERSION) return __put_user(SOUND_VERSION, (int __user *)p); + lock_kernel(); if (_IOC_TYPE(cmd) == 'M' && num_mixers > 0 && /* Mixer ioctl */ (dev & 0x0f) != SND_DEV_CTL) { dtype = dev & 0x0f; @@ -360,24 +361,31 @@ static int sound_ioctl(struct inode *inode, struct file *file, case SND_DEV_DSP: case SND_DEV_DSP16: case SND_DEV_AUDIO: - return sound_mixer_ioctl(audio_devs[dev >> 4]->mixer_dev, + ret = sound_mixer_ioctl(audio_devs[dev >> 4]->mixer_dev, cmd, p); - + break; default: - return sound_mixer_ioctl(dev >> 4, cmd, p); + ret = sound_mixer_ioctl(dev >> 4, cmd, p); + break; } + unlock_kernel(); + return ret; } + switch (dev & 0x0f) { case SND_DEV_CTL: if (cmd == SOUND_MIXER_GETLEVELS) - return get_mixer_levels(p); - if (cmd == SOUND_MIXER_SETLEVELS) - return set_mixer_levels(p); - return sound_mixer_ioctl(dev >> 4, cmd, p); + ret = get_mixer_levels(p); + else if (cmd == SOUND_MIXER_SETLEVELS) + ret = set_mixer_levels(p); + else + ret = sound_mixer_ioctl(dev >> 4, cmd, p); + break; case SND_DEV_SEQ: case SND_DEV_SEQ2: - return sequencer_ioctl(dev, file, cmd, p); + ret = sequencer_ioctl(dev, file, cmd, p); + break; case SND_DEV_DSP: case SND_DEV_DSP16: @@ -390,7 +398,8 @@ static int sound_ioctl(struct inode *inode, struct file *file, break; } - return -EINVAL; + unlock_kernel(); + return ret; } static unsigned int sound_poll(struct file *file, poll_table * wait) @@ -490,7 +499,7 @@ const struct file_operations oss_sound_fops = { .read = sound_read, .write = sound_write, .poll = sound_poll, - .ioctl = sound_ioctl, + .unlocked_ioctl = sound_ioctl, .mmap = sound_mmap, .open = sound_open, .release = sound_release, -- cgit v0.10.2 From 03e7a35c0ef7a462385fb6a301dfc1b287cac6de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Jan 2010 14:01:19 +0000 Subject: Revert "ASoC: ad1836: reset and restore clock control mode in suspend/resume entry" This reverts commit afe1c2cd71eb4e0fade720b5709722e7124f29c0 since it doesn't build. diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 83add2f..2c18e3d 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,36 +223,6 @@ static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, return reg_cache[reg]; } -#ifdef CONFIG_PM -static int ad1836_soc_suspend(struct platform_device *pdev, - pm_message_t state) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - /* reset clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} - -static int ad1836_soc_resume(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - /* restore clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 |= AD1836_ADC_AUX; - - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} -#else -#define ad1836_soc_suspend NULL -#define ad1836_soc_resume NULL -#endif - static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -434,8 +404,6 @@ static int ad1836_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, - .suspend = ad1836_soc_suspend, - .resume = ad1836_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); -- cgit v0.10.2 From 735fe4cfbc3cedea41bd0ed31955054dae6beb46 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Jan 2010 14:13:00 +0000 Subject: ASoC: Add missing __devexit and __devinit annotations Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index fbf3ab4..cf2975a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -471,8 +471,8 @@ init_err: } #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static int da7210_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int __devinit da7210_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct da7210_priv *da7210; struct snd_soc_codec *codec; @@ -495,7 +495,7 @@ static int da7210_i2c_probe(struct i2c_client *i2c, return ret; } -static int da7210_i2c_remove(struct i2c_client *client) +static int __devexit da7210_i2c_remove(struct i2c_client *client) { struct da7210_priv *da7210 = i2c_get_clientdata(client); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 3ef3255..2df9c20 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1191,8 +1191,8 @@ struct snd_soc_dai dac33_dai = { }; EXPORT_SYMBOL_GPL(dac33_dai); -static int dac33_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) +static int __devinit dac33_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) { struct tlv320dac33_platform_data *pdata; struct tlv320dac33_priv *dac33; @@ -1345,7 +1345,7 @@ error_reg: return ret; } -static int dac33_i2c_remove(struct i2c_client *client) +static int __devexit dac33_i2c_remove(struct i2c_client *client) { struct tlv320dac33_priv *dac33; diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8b27281..958d49c 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -379,8 +379,8 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); -static int tpa6130a2_probe(struct i2c_client *client, - const struct i2c_device_id *id) +static int __devinit tpa6130a2_probe(struct i2c_client *client, + const struct i2c_device_id *id) { struct device *dev; struct tpa6130a2_data *data; @@ -479,7 +479,7 @@ err_gpio: return ret; } -static int tpa6130a2_remove(struct i2c_client *client) +static int __devexit tpa6130a2_remove(struct i2c_client *client) { struct tpa6130a2_data *data = i2c_get_clientdata(client); -- cgit v0.10.2 From ed69c6a8eef679f2783848ed624897a937a434ac Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 13 Jan 2010 08:12:31 +0100 Subject: ALSA: pcm_lib - fix wrong delta print for jiffies check The previous jiffies delta was 0 in all cases. Use hw_ptr variable to store and print original value. Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0ee7e80..5417f7d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -394,6 +394,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, + HZ/100); /* move new_hw_ptr according jiffies not pos variable */ new_hw_ptr = old_hw_ptr; + hw_base = delta; /* use loop to avoid checks for delta overflows */ /* the delta value is small or zero in most cases */ while (delta > 0) { @@ -403,8 +404,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, delta--; } /* align hw_base to buffer_size */ - hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); - delta = 0; hw_ptr_error(substream, "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " @@ -412,9 +411,12 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, - ((hdelta * HZ) / runtime->rate), delta, + ((hdelta * HZ) / runtime->rate), hw_base, (unsigned long)old_hw_ptr, (unsigned long)new_hw_ptr); + /* reset values to proper state */ + delta = 0; + hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { -- cgit v0.10.2 From d2f2fcd2541bae004db7f4798ffd9d2cb75ae817 Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Tue, 12 Jan 2010 17:03:35 -0800 Subject: ALSA: hda_intel: ALSA HD Audio patch for Intel Cougar Point DeviceIDs This patch adds the Intel Cougar Point (PCH) HD Audio Controller DeviceIDs. Signed-off-by: Seth Heasley Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1f516e6..6d331c4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -125,6 +125,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH9}," "{Intel, ICH10}," "{Intel, PCH}," + "{Intel, CPT}," "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," @@ -2677,6 +2678,8 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, /* PCH */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, + /* CPT */ + { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_ICH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */ -- cgit v0.10.2 From fd63df2264f2518fa67dca596d493a330537494d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 13 Jan 2010 12:37:49 +0200 Subject: ASoC: TWL4030: Replace comma with semicolon in probe function The codec structure initialization statements should be separated by semicolons. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a27f7b..74f0d65 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2192,7 +2192,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->num_dai = ARRAY_SIZE(twl4030_dai); codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); -- cgit v0.10.2 From 617b14c50eb95b36360b2b3232c6cf20b910e2f8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 13 Jan 2010 11:25:05 +0100 Subject: ASoC: ak4104: allow more sample rates The transmitter supports all sample rates up to 192KHz, so the driver should not give a limit. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 3a14c6f..b9ef7e4 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -185,9 +185,7 @@ struct snd_soc_dai ak4104_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_32000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE -- cgit v0.10.2 From d1458279bf9c575a52fd22818ca19c463f380aba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 09:16:52 +0100 Subject: ALSA: Add snd_pci_quirk_lookup_id() Added a new function to look up a quirk entry with the given PCI SSID instead of a pci device pointer. This can be used when the searched ID is overridden for debugging or such a purpose. Signed-off-by: Takashi Iwai diff --git a/include/sound/core.h b/include/sound/core.h index a61499c..89e0ac1 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -458,5 +458,8 @@ struct snd_pci_quirk { const struct snd_pci_quirk * snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list); +const struct snd_pci_quirk * +snd_pci_quirk_lookup_id(u16 vendor, u16 device, + const struct snd_pci_quirk *list); #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 23a032c..3da4f92 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -101,8 +101,9 @@ EXPORT_SYMBOL_GPL(__snd_printk); #ifdef CONFIG_PCI #include /** - * snd_pci_quirk_lookup - look up a PCI SSID quirk list - * @pci: pci_dev handle + * snd_pci_quirk_lookup_id - look up a PCI SSID quirk list + * @vendor: PCI SSV id + * @device: PCI SSD id * @list: quirk list, terminated by a null entry * * Look through the given quirk list and finds a matching entry @@ -112,18 +113,39 @@ EXPORT_SYMBOL_GPL(__snd_printk); * Returns the matched entry pointer, or NULL if nothing matched. */ const struct snd_pci_quirk * -snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) +snd_pci_quirk_lookup_id(u16 vendor, u16 device, + const struct snd_pci_quirk *list) { const struct snd_pci_quirk *q; for (q = list; q->subvendor; q++) { - if (q->subvendor != pci->subsystem_vendor) + if (q->subvendor != vendor) continue; if (!q->subdevice || - (pci->subsystem_device & q->subdevice_mask) == q->subdevice) + (device & q->subdevice_mask) == q->subdevice) return q; } return NULL; } +EXPORT_SYMBOL(snd_pci_quirk_lookup_id); + +/** + * snd_pci_quirk_lookup - look up a PCI SSID quirk list + * @pci: pci_dev handle + * @list: quirk list, terminated by a null entry + * + * Look through the given quirk list and finds a matching entry + * with the same PCI SSID. When subdevice is 0, all subdevice + * values may match. + * + * Returns the matched entry pointer, or NULL if nothing matched. + */ +const struct snd_pci_quirk * +snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) +{ + return snd_pci_quirk_lookup_id(pci->subsystem_vendor, + pci->subsystem_device, + list); +} EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif -- cgit v0.10.2 From 408bffd01cfcda2907b07fb86b3666e3db86fd82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 09:19:46 +0100 Subject: ALSA: ctxfi - Add subsystem option Added a new option "subsystem" to override the PCI SSID for identifying the card type. Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index c540637..c83fd7b 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -482,6 +482,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. reference_rate - reference sample rate, 44100 or 48000 (default) multiple - multiple to ref. sample rate, 1 or 2 (default) + subsystem - override the PCI SSID for probing; the value + consists of SSVID << 16 | SSDID. The default is + zero, which means no override. This module supports multiple cards. diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index cb65bd0..903594e 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1225,10 +1225,11 @@ static int atc_dev_free(struct snd_device *dev) return ct_atc_destroy(atc); } -static int __devinit atc_identify_card(struct ct_atc *atc) +static int __devinit atc_identify_card(struct ct_atc *atc, unsigned int ssid) { const struct snd_pci_quirk *p; const struct snd_pci_quirk *list; + u16 vendor_id, device_id; switch (atc->chip_type) { case ATC20K1: @@ -1242,13 +1243,19 @@ static int __devinit atc_identify_card(struct ct_atc *atc) default: return -ENOENT; } - p = snd_pci_quirk_lookup(atc->pci, list); + if (ssid) { + vendor_id = ssid >> 16; + device_id = ssid & 0xffff; + } else { + vendor_id = atc->pci->subsystem_vendor; + device_id = atc->pci->subsystem_device; + } + p = snd_pci_quirk_lookup_id(vendor_id, device_id, list); if (p) { if (p->value < 0) { printk(KERN_ERR "ctxfi: " "Device %04x:%04x is black-listed\n", - atc->pci->subsystem_vendor, - atc->pci->subsystem_device); + vendor_id, device_id); return -ENOENT; } atc->model = p->value; @@ -1261,8 +1268,7 @@ static int __devinit atc_identify_card(struct ct_atc *atc) atc->model_name = ct_subsys_name[atc->model]; snd_printd("ctxfi: chip %s model %s (%04x:%04x) is found\n", atc->chip_name, atc->model_name, - atc->pci->subsystem_vendor, - atc->pci->subsystem_device); + vendor_id, device_id); return 0; } @@ -1636,7 +1642,8 @@ static struct ct_atc atc_preset __devinitdata = { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, - int chip_type, struct ct_atc **ratc) + int chip_type, unsigned int ssid, + struct ct_atc **ratc) { struct ct_atc *atc; static struct snd_device_ops ops = { @@ -1662,7 +1669,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, mutex_init(&atc->atc_mutex); /* Find card model */ - err = atc_identify_card(atc); + err = atc_identify_card(atc, ssid); if (err < 0) { printk(KERN_ERR "ctatc: Card not recognised\n"); goto error1; diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 9fd8a57..7167c01 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -148,7 +148,7 @@ struct ct_atc { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, int chip_type, - struct ct_atc **ratc); + unsigned int subsysid, struct ct_atc **ratc); int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc); #endif /* CTATC_H */ diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 7654174..ed44ed7 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -32,6 +32,7 @@ module_param(multiple, uint, S_IRUGO); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static unsigned int subsystem[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Creative X-Fi driver"); @@ -39,6 +40,8 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Creative X-Fi driver"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); +module_param_array(subsystem, int, NULL, 0444); +MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); static struct pci_device_id ct_pci_dev_ids[] = { /* only X-Fi is supported, so... */ @@ -85,7 +88,7 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) multiple = 2; } err = ct_atc_create(card, pci, reference_rate, multiple, - pci_id->driver_data, &atc); + pci_id->driver_data, subsystem[dev], &atc); if (err < 0) goto error; -- cgit v0.10.2 From 738ada47cf60830d37bb70ffb0b0281d19fc4c7f Mon Sep 17 00:00:00 2001 From: Thomas Weber Date: Tue, 12 Jan 2010 17:07:18 +0100 Subject: ASoC: TWL4030: Fix typo in comment in header file Signed-off-by: Thomas Weber Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index dd6396e..f206d24 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -25,7 +25,7 @@ /* Register descriptions are here */ #include -/* Sgadow register used by the audio driver */ +/* Shadow register used by the audio driver */ #define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) -- cgit v0.10.2 From 6aababdf20bb8892023bb8df136514d7679e4959 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 15 Jan 2010 17:36:48 +0100 Subject: ASoC: cs4270: allow passing freq=0 in set_dai_sysclk() For setups with variable MCLKs, the current logic of limiting the available sampling rates at startup time is not sufficient. We need to be able to change the setting at a later point, and so the codec must offer all possible rates until the hw_params are given. This patches allows that by passing 0 as 'freq' argument to cs4270_set_dai_sysclk(). Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8b54575..593bfc7 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -200,6 +200,11 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = { * This function must be called by the machine driver's 'startup' function, * otherwise the list of supported sample rates will not be available in * time for ALSA. + * + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause + * theoretically possible sample rates to be enabled. Call it again with a + * proper value set one the external clock is set (most probably you would do + * that from a machine's driver 'hw_param' hook. */ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) @@ -213,20 +218,27 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, cs4270->mclk = freq; - for (i = 0; i < NUM_MCLK_RATIOS; i++) { - unsigned int rate = freq / cs4270_mode_ratios[i].ratio; - rates |= snd_pcm_rate_to_rate_bit(rate); - if (rate < rate_min) - rate_min = rate; - if (rate > rate_max) - rate_max = rate; - } - /* FIXME: soc should support a rate list */ - rates &= ~SNDRV_PCM_RATE_KNOT; + if (cs4270->mclk) { + for (i = 0; i < NUM_MCLK_RATIOS; i++) { + unsigned int rate = freq / cs4270_mode_ratios[i].ratio; + rates |= snd_pcm_rate_to_rate_bit(rate); + if (rate < rate_min) + rate_min = rate; + if (rate > rate_max) + rate_max = rate; + } + /* FIXME: soc should support a rate list */ + rates &= ~SNDRV_PCM_RATE_KNOT; - if (!rates) { - dev_err(codec->dev, "could not find a valid sample rate\n"); - return -EINVAL; + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + } else { + /* enable all possible rates */ + rates = SNDRV_PCM_RATE_8000_192000; + rate_min = 8000; + rate_max = 192000; } codec_dai->playback.rates = rates; -- cgit v0.10.2 From a421296840379aee7d00ec4a28ecfe7e697a0a44 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 15 Jan 2010 17:36:49 +0100 Subject: ASoC: support more sample rates on raumfeld devices Add support for sample rates other than 44100Khz on raumfeld audio devices. At startup time, call snd_soc_dai_set_sysclk() with 0 as 'freq' argument so it offers all the sample rates. Later, the function is called again to give proper constraints. Use the external audio clock generator to provide double data rate clocks as the PXA's internal baud generator does anything but what's described in the datasheets. Signed-off-by: Daniel Mack Cc: Mark Brown Cc: Timur Tabi Signed-off-by: Mark Brown diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index acfce1c..7e3f416 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -41,7 +41,9 @@ static struct i2c_board_info max9486_hwmon_info = { }; #define MAX9485_MCLK_FREQ_112896 0x22 -#define MAX9485_MCLK_FREQ_122880 0x23 +#define MAX9485_MCLK_FREQ_122880 0x23 +#define MAX9485_MCLK_FREQ_225792 0x32 +#define MAX9485_MCLK_FREQ_245760 0x33 static void set_max9485_clk(char clk) { @@ -71,9 +73,17 @@ static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - set_max9485_clk(MAX9485_MCLK_FREQ_112896); + /* set freq to 0 to enable all possible codec sample rates */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); +} - return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* set freq to 0 to enable all possible codec sample rates */ + snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); } static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, @@ -86,20 +96,24 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, int ret = 0; switch (params_rate(params)) { - case 8000: - case 16000: + case 44100: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; case 48000: - case 96000: set_max9485_clk(MAX9485_MCLK_FREQ_122880); clk = 12288000; break; - case 11025: - case 22050: - case 44100: case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; + set_max9485_clk(MAX9485_MCLK_FREQ_225792); + clk = 22579200; break; + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_245760); + clk = 24576000; + break; + default: + return -EINVAL; } fmt = SND_SOC_DAIFMT_I2S | @@ -128,7 +142,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -137,6 +151,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, static struct snd_soc_ops raumfeld_cs4270_ops = { .startup = raumfeld_cs4270_startup, + .shutdown = raumfeld_cs4270_shutdown, .hw_params = raumfeld_cs4270_hw_params, }; @@ -181,20 +196,24 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, int fmt, ret = 0, clk = 0; switch (params_rate(params)) { - case 8000: - case 16000: + case 44100: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; case 48000: - case 96000: set_max9485_clk(MAX9485_MCLK_FREQ_122880); clk = 12288000; break; - case 11025: - case 22050: - case 44100: case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; + set_max9485_clk(MAX9485_MCLK_FREQ_225792); + clk = 22579200; + break; + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_245760); + clk = 24576000; break; + default: + return -EINVAL; } fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; @@ -217,7 +236,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; -- cgit v0.10.2 From 8380222ec9458d38a4e0cc3cb688ad7fff311df4 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Wed, 25 Nov 2009 16:41:04 +0100 Subject: ASoC: Add a new imx-ssi sound driver The old driver has the number of SSI units in the system hardcoded, does not make use of the device model and works only on i.MX21/27. This driver replaces it. It works in DMA mode on i.MX21/27 and using an FIQ handler on other systems. It also supports AC97 mode of the SSI units. Signed-off-by: Sascha Hauer Acked-by: Javier Martin Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/arch/arm/plat-mxc/Makefile b/arch/arm/plat-mxc/Makefile index e3212c8..b0b9fc3 100644 --- a/arch/arm/plat-mxc/Makefile +++ b/arch/arm/plat-mxc/Makefile @@ -9,3 +9,9 @@ obj-$(CONFIG_ARCH_MX1) += iomux-mx1-mx2.o dma-mx1-mx2.o obj-$(CONFIG_ARCH_MX2) += iomux-mx1-mx2.o dma-mx1-mx2.o obj-$(CONFIG_ARCH_MXC_IOMUX_V3) += iomux-v3.o obj-$(CONFIG_MXC_PWM) += pwm.o +obj-$(CONFIG_ARCH_MXC_AUDMUX_V1) += audmux-v1.o +obj-$(CONFIG_ARCH_MXC_AUDMUX_V2) += audmux-v2.o +ifdef CONFIG_SND_IMX_SOC +obj-y += ssi-fiq.o +obj-y += ssi-fiq-ksym.o +endif diff --git a/arch/arm/plat-mxc/ssi-fiq-ksym.c b/arch/arm/plat-mxc/ssi-fiq-ksym.c new file mode 100644 index 0000000..b5fad45 --- /dev/null +++ b/arch/arm/plat-mxc/ssi-fiq-ksym.c @@ -0,0 +1,20 @@ +/* + * Exported ksyms for the SSI FIQ handler + * + * Copyright (C) 2009, Sascha Hauer + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include + +#include + +EXPORT_SYMBOL(imx_ssi_fiq_tx_buffer); +EXPORT_SYMBOL(imx_ssi_fiq_rx_buffer); +EXPORT_SYMBOL(imx_ssi_fiq_start); +EXPORT_SYMBOL(imx_ssi_fiq_end); +EXPORT_SYMBOL(imx_ssi_fiq_base); + diff --git a/arch/arm/plat-mxc/ssi-fiq.S b/arch/arm/plat-mxc/ssi-fiq.S new file mode 100644 index 0000000..4ddce56 --- /dev/null +++ b/arch/arm/plat-mxc/ssi-fiq.S @@ -0,0 +1,134 @@ +/* + * Copyright (C) 2009 Sascha Hauer + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include + +/* + * r8 = bit 0-15: tx offset, bit 16-31: tx buffer size + * r9 = bit 0-15: rx offset, bit 16-31: rx buffer size + */ + +#define SSI_STX0 0x00 +#define SSI_SRX0 0x08 +#define SSI_SISR 0x14 +#define SSI_SIER 0x18 +#define SSI_SACNT 0x38 + +#define SSI_SACNT_AC97EN (1 << 0) + +#define SSI_SIER_TFE0_EN (1 << 0) +#define SSI_SISR_TFE0 (1 << 0) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SIER_RFF0_EN (1 << 2) + + .text + .global imx_ssi_fiq_start + .global imx_ssi_fiq_end + .global imx_ssi_fiq_base + .global imx_ssi_fiq_rx_buffer + .global imx_ssi_fiq_tx_buffer + +imx_ssi_fiq_start: + ldr r12, imx_ssi_fiq_base + + /* TX */ + ldr r11, imx_ssi_fiq_tx_buffer + + /* shall we send? */ + ldr r13, [r12, #SSI_SIER] + tst r13, #SSI_SIER_TFE0_EN + beq 1f + + /* TX FIFO empty? */ + ldr r13, [r12, #SSI_SISR] + tst r13, #SSI_SISR_TFE0 + beq 1f + + mov r10, #0x10000 + sub r10, #1 + and r10, r10, r8 /* r10: current buffer offset */ + + add r11, r11, r10 + + ldrh r13, [r11] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #2] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #4] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #6] + strh r13, [r12, #SSI_STX0] + + add r10, #8 + lsr r13, r8, #16 /* r13: buffer size */ + cmp r10, r13 + lslgt r8, r13, #16 + addle r8, #8 +1: + /* RX */ + + /* shall we receive? */ + ldr r13, [r12, #SSI_SIER] + tst r13, #SSI_SIER_RFF0_EN + beq 1f + + /* RX FIFO full? */ + ldr r13, [r12, #SSI_SISR] + tst r13, #SSI_SISR_RFF0 + beq 1f + + ldr r11, imx_ssi_fiq_rx_buffer + + mov r10, #0x10000 + sub r10, #1 + and r10, r10, r9 /* r10: current buffer offset */ + + add r11, r11, r10 + + ldr r13, [r12, #SSI_SACNT] + tst r13, #SSI_SACNT_AC97EN + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #2] + + /* dummy read to skip slot 12 */ + ldrne r13, [r12, #SSI_SRX0] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #4] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #6] + + /* dummy read to skip slot 12 */ + ldrne r13, [r12, #SSI_SRX0] + + add r10, #8 + lsr r13, r9, #16 /* r13: buffer size */ + cmp r10, r13 + lslgt r9, r13, #16 + addle r9, #8 + +1: + @ return from FIQ + subs pc, lr, #4 +imx_ssi_fiq_base: + .word 0x0 +imx_ssi_fiq_rx_buffer: + .word 0x0 +imx_ssi_fiq_tx_buffer: + .word 0x0 +imx_ssi_fiq_end: + diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index a700562e..84a25e6 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,21 +1,13 @@ -config SND_MX1_MX2_SOC - tristate "SoC Audio for Freecale i.MX1x i.MX2x CPUs" - depends on ARCH_MX2 || ARCH_MX1 +config SND_IMX_SOC + tristate "SoC Audio for Freecale i.MX CPUs" + depends on ARCH_MXC select SND_PCM + select FIQ + select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to - the MX1 or MX2 SSI interface. + the i.MX SSI interface. config SND_MXC_SOC_SSI tristate -config SND_SOC_MX27VIS_WM8974 - tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board" - depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10 - select SND_MXC_SOC_SSI - select SND_SOC_WM8974 - help - Say Y if you want to add support for SoC audio on Visstrim SM10 - board with WM8974. - - diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index c2ffd2c..4bde34a 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,10 +1,10 @@ # i.MX Platform Support -snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o -snd-soc-mxc-ssi-objs := mxc-ssi.o +snd-soc-imx-objs := imx-ssi.o imx-pcm-fiq.o imx-pcm-dma-mx2.o -obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o -obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o +ifdef CONFIG_MACH_MX27 +snd-soc-imx-objs += imx-pcm-dma-mx2.o +endif + +obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support -snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o -obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c new file mode 100644 index 0000000..19452e4 --- /dev/null +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -0,0 +1,313 @@ +/* + * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int sg_count; + struct scatterlist *sg_list; + int period; + int periods; + unsigned long dma_addr; + int dma; + struct snd_pcm_substream *substream; + unsigned long offset; + unsigned long size; + unsigned long period_cnt; + void *buf; + int period_time; +}; + +/* Called by the DMA framework when a period has elapsed */ +static void imx_ssi_dma_progression(int channel, void *data, + struct scatterlist *sg) +{ + struct snd_pcm_substream *substream = data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + if (!sg) + return; + + runtime = iprtd->substream->runtime; + + iprtd->offset = sg->dma_address - runtime->dma_addr; + + snd_pcm_period_elapsed(iprtd->substream); +} + +static void imx_ssi_dma_callback(int channel, void *data) +{ + pr_err("%s shouldn't be called\n", __func__); +} + +static void snd_imx_dma_err_callback(int channel, void *data, int err) +{ + pr_err("DMA error callback called\n"); + + pr_err("DMA timeout on channel %d -%s%s%s%s\n", + channel, + err & IMX_DMA_ERR_BURST ? " burst" : "", + err & IMX_DMA_ERR_REQUEST ? " request" : "", + err & IMX_DMA_ERR_TRANSFER ? " transfer" : "", + err & IMX_DMA_ERR_BUFFER ? " buffer" : ""); +} + +static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int ret; + + iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); + if (iprtd->dma < 0) { + pr_err("Failed to claim the audio DMA\n"); + return -ENODEV; + } + + ret = imx_dma_setup_handlers(iprtd->dma, + imx_ssi_dma_callback, + snd_imx_dma_err_callback, substream); + if (ret) + goto out; + + ret = imx_dma_setup_progression_handler(iprtd->dma, + imx_ssi_dma_progression); + if (ret) { + pr_err("Failed to setup the DMA handler\n"); + goto out; + } + + ret = imx_dma_config_channel(iprtd->dma, + IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, + dma_params->dma, 1); + if (ret < 0) { + pr_err("Cannot configure DMA channel: %d\n", ret); + goto out; + } + + imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2); + + return 0; +out: + imx_dma_free(iprtd->dma); + return ret; +} + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int i; + unsigned long dma_addr; + + imx_ssi_dma_alloc(substream); + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / + params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + if (iprtd->sg_count != iprtd->periods) { + kfree(iprtd->sg_list); + + iprtd->sg_list = kcalloc(iprtd->periods + 1, + sizeof(struct scatterlist), GFP_KERNEL); + if (!iprtd->sg_list) + return -ENOMEM; + iprtd->sg_count = iprtd->periods + 1; + } + + sg_init_table(iprtd->sg_list, iprtd->sg_count); + dma_addr = runtime->dma_addr; + + for (i = 0; i < iprtd->periods; i++) { + iprtd->sg_list[i].page_link = 0; + iprtd->sg_list[i].offset = 0; + iprtd->sg_list[i].dma_address = dma_addr; + iprtd->sg_list[i].length = iprtd->period; + dma_addr += iprtd->period; + } + + /* close the loop */ + iprtd->sg_list[iprtd->sg_count - 1].offset = 0; + iprtd->sg_list[iprtd->sg_count - 1].length = 0; + iprtd->sg_list[iprtd->sg_count - 1].page_link = + ((unsigned long) iprtd->sg_list | 0x01) & ~0x02; + return 0; +} + +static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + if (iprtd->dma >= 0) { + imx_dma_free(iprtd->dma); + iprtd->dma = -EINVAL; + } + + kfree(iprtd->sg_list); + iprtd->sg_list = NULL; + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int err; + + iprtd->substream = substream; + iprtd->buf = (unsigned int *)substream->dma_buffer.area; + iprtd->period_cnt = 0; + + pr_debug("%s: buf: %p period: %d periods: %d\n", + __func__, iprtd->buf, iprtd->period, iprtd->periods); + + err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, + IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_MODE_WRITE : DMA_MODE_READ); + if (err) + return err; + + return 0; +} + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + imx_dma_enable(iprtd->dma); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + imx_dma_disable(iprtd->dma); + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + runtime->private_data = iprtd; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .hw_free = snd_imx_pcm_hw_free, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static struct snd_soc_platform imx_soc_platform_dma = { + .name = "imx-audio", + .pcm_ops = &imx_pcm_ops, + .pcm_new = imx_pcm_new, + .pcm_free = imx_pcm_free, +}; + +struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + ssi->dma_params_tx.burstsize = DMA_TXFIFO_BURST; + ssi->dma_params_rx.burstsize = DMA_RXFIFO_BURST; + + return &imx_soc_platform_dma; +} + diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c new file mode 100644 index 0000000..5532579 --- /dev/null +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -0,0 +1,277 @@ +/* + * imx-pcm-fiq.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include + +#include + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int period; + int periods; + unsigned long dma_addr; + int dma; + unsigned long offset; + unsigned long size; + unsigned long period_cnt; + void *buf; + struct timer_list timer; + int period_time; +}; + +static void imx_ssi_timer_callback(unsigned long data) +{ + struct snd_pcm_substream *substream = (void *)data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(®s); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + iprtd->offset = regs.ARM_r8 & 0xffff; + else + iprtd->offset = regs.ARM_r9 & 0xffff; + + iprtd->timer.expires = jiffies + iprtd->period_time; + add_timer(&iprtd->timer); + snd_pcm_period_elapsed(substream); +} + +static struct fiq_handler fh = { + .name = DRV_NAME, +}; + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(®s); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regs.ARM_r8 = (iprtd->period * iprtd->periods - 1) << 16; + else + regs.ARM_r9 = (iprtd->period * iprtd->periods - 1) << 16; + + set_fiq_regs(®s); + + return 0; +} + +static int fiq_enable; +static int imx_pcm_fiq; + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + iprtd->timer.expires = jiffies + iprtd->period_time; + add_timer(&iprtd->timer); + if (++fiq_enable == 1) + enable_fiq(imx_pcm_fiq); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + del_timer(&iprtd->timer); + if (--fiq_enable == 0) + disable_fiq(imx_pcm_fiq); + + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + runtime->private_data = iprtd; + + init_timer(&iprtd->timer); + iprtd->timer.data = (unsigned long)substream; + iprtd->timer.function = imx_ssi_timer_callback; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static int snd_imx_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + del_timer_sync(&iprtd->timer); + kfree(iprtd); + + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .close = snd_imx_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret; + + ret = imx_pcm_new(card, dai, pcm); + if (ret) + return ret; + + if (dai->playback.channels_min) { + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; + } + + if (dai->capture.channels_min) { + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; + } + + set_fiq_handler(&imx_ssi_fiq_start, + &imx_ssi_fiq_end - &imx_ssi_fiq_start); + + return 0; +} + +static struct snd_soc_platform imx_soc_platform_fiq = { + .pcm_ops = &imx_pcm_ops, + .pcm_new = imx_pcm_fiq_new, + .pcm_free = imx_pcm_free, +}; + +struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + int ret = 0; + + ret = claim_fiq(&fh); + if (ret) { + dev_err(&pdev->dev, "failed to claim fiq: %d", ret); + return ERR_PTR(ret); + } + + mxc_set_irq_fiq(ssi->irq, 1); + + imx_pcm_fiq = ssi->irq; + + imx_ssi_fiq_base = (unsigned long)ssi->base; + + ssi->dma_params_tx.burstsize = 4; + ssi->dma_params_rx.burstsize = 6; + + return &imx_soc_platform_fiq; +} + +void imx_ssi_fiq_exit(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + mxc_set_irq_fiq(ssi->irq, 0); + release_fiq(&fh); +} + diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c new file mode 100644 index 0000000..c57a11f --- /dev/null +++ b/sound/soc/imx/imx-ssi.c @@ -0,0 +1,762 @@ +/* + * imx-ssi.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * + * The i.MX SSI core has some nasty limitations in AC97 mode. While most + * sane processor vendors have a FIFO per AC97 slot, the i.MX has only + * one FIFO which combines all valid receive slots. We cannot even select + * which slots we want to receive. The WM9712 with which this driver + * was developped with always sends GPIO status data in slot 12 which + * we receive in our (PCM-) data stream. The only chance we have is to + * manually skip this data in the FIQ handler. With sampling rates different + * from 48000Hz not every frame has valid receive data, so the ratio + * between pcm data and GPIO status data changes. Our FIQ handler is not + * able to handle this, hence this driver only works with 48000Hz sampling + * rate. + * Reading and writing AC97 registers is another challange. The core + * provides us status bits when the read register is updated with *another* + * value. When we read the same register two times (and the register still + * contains the same value) these status bits are not set. We work + * around this by not polling these bits but only wait a fixed delay. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "imx-ssi.h" + +#define SSI_SACNT_DEFAULT (SSI_SACNT_AC97EN | SSI_SACNT_FV) + +/* + * SSI Network Mode or TDM slots configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 sccr; + + sccr = readl(ssi->base + SSI_STCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_STCCR); + + sccr = readl(ssi->base + SSI_SRCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_SRCCR); + + writel(tx_mask, ssi->base + SSI_STMSK); + writel(rx_mask, ssi->base + SSI_SRMSK); + + return 0; +} + +/* + * SSI DAI format configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + * Note: We don't use the I2S modes but instead manually configure the + * SSI for I2S because the I2S mode is only a register preset. + */ +static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 strcr = 0, scr; + + scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET); + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data on rising edge of bclk, frame low 1clk before data */ + strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + scr |= SSI_SCR_NET; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TXBIT0; + break; + case SND_SOC_DAIFMT_DSP_B: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TFSL; + break; + case SND_SOC_DAIFMT_DSP_A: + /* data on rising edge of bclk, frame high 1clk before data */ + strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + break; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + strcr |= SSI_STCR_TFSI; + strcr &= ~SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_IB_NF: + strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + break; + case SND_SOC_DAIFMT_NB_IF: + strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_NF: + strcr &= ~SSI_STCR_TFSI; + strcr |= SSI_STCR_TSCKP; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr |= SSI_STCR_TFDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + strcr |= SSI_STCR_TXDIR; + break; + } + + strcr |= SSI_STCR_TFEN0; + + writel(strcr, ssi->base + SSI_STCR); + writel(strcr, ssi->base + SSI_SRCR); + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI system clock configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 scr; + + scr = readl(ssi->base + SSI_SCR); + + switch (clk_id) { + case IMX_SSP_SYS_CLK: + if (dir == SND_SOC_CLOCK_OUT) + scr |= SSI_SCR_SYS_CLK_EN; + else + scr &= ~SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI Clock dividers + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 stccr, srccr; + + stccr = readl(ssi->base + SSI_STCCR); + srccr = readl(ssi->base + SSI_SRCCR); + + switch (div_id) { + case IMX_SSI_TX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + case IMX_SSI_RX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + default: + return -EINVAL; + } + + writel(stccr, ssi->base + SSI_STCCR); + writel(srccr, ssi->base + SSI_SRCCR); + + return 0; +} + +/* + * Should only be called when port is inactive (i.e. SSIEN = 0), + * although can be called multiple times by upper layers. + */ +static int imx_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 reg, sccr; + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg = SSI_STCCR; + cpu_dai->dma_data = &ssi->dma_params_tx; + } else { + reg = SSI_SRCCR; + cpu_dai->dma_data = &ssi->dma_params_rx; + } + + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + sccr |= SSI_SRCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sccr |= SSI_SRCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + sccr |= SSI_SRCCR_WL(24); + break; + } + + writel(sccr, ssi->base + reg); + + return 0; +} + +static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + unsigned int sier_bits, sier; + unsigned int scr; + + scr = readl(ssi->base + SSI_SCR); + sier = readl(ssi->base + SSI_SIER); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_TDMAE; + else + sier_bits = SSI_SIER_TIE | SSI_SIER_TFE0_EN; + } else { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_RDMAE; + else + sier_bits = SSI_SIER_RIE | SSI_SIER_RFF0_EN; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr |= SSI_SCR_TE; + else + scr |= SSI_SCR_RE; + sier |= sier_bits; + + if (++ssi->enabled == 1) + scr |= SSI_SCR_SSIEN; + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr &= ~SSI_SCR_TE; + else + scr &= ~SSI_SCR_RE; + sier &= ~sier_bits; + + if (--ssi->enabled == 0) + scr &= ~SSI_SCR_SSIEN; + + break; + default: + return -EINVAL; + } + + if (!(ssi->flags & IMX_SSI_USE_AC97)) + /* rx/tx are always enabled to access ac97 registers */ + writel(scr, ssi->base + SSI_SCR); + + writel(sier, ssi->base + SSI_SIER); + + return 0; +} + +static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .hw_params = imx_ssi_hw_params, + .set_fmt = imx_ssi_set_dai_fmt, + .set_clkdiv = imx_ssi_set_dai_clkdiv, + .set_sysclk = imx_ssi_set_dai_sysclk, + .set_tdm_slot = imx_ssi_set_dai_tdm_slot, + .trigger = imx_ssi_trigger, +}; + +static struct snd_soc_dai imx_ssi_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); + + pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + return ret; +} + +static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = IMX_SSI_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); + +int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &imx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + if (dai->playback.channels_min) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + +void imx_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +struct snd_soc_platform imx_soc_platform = { + .name = "imx-audio", +}; +EXPORT_SYMBOL_GPL(imx_soc_platform); + +static struct snd_soc_dai imx_ac97_dai = { + .name = "AC97", + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +static void setup_channel_to_ac97(struct imx_ssi *imx_ssi) +{ + void __iomem *base = imx_ssi->base; + + writel(0x0, base + SSI_SCR); + writel(0x0, base + SSI_STCR); + writel(0x0, base + SSI_SRCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET, base + SSI_SCR); + + writel(SSI_SFCSR_RFWM0(8) | + SSI_SFCSR_TFWM0(8) | + SSI_SFCSR_RFWM1(8) | + SSI_SFCSR_TFWM1(8), base + SSI_SFCSR); + + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_STCCR); + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_SRCCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN, base + SSI_SCR); + writel(SSI_SOR_WAIT(3), base + SSI_SOR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN | + SSI_SCR_TE | SSI_SCR_RE, + base + SSI_SCR); + + writel(SSI_SACNT_DEFAULT, base + SSI_SACNT); + writel(0xff, base + SSI_SACCDIS); + writel(0x300, base + SSI_SACCEN); +} + +static struct imx_ssi *ac97_ssi; + +static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + unsigned int lreg; + unsigned int lval; + + if (reg > 0x7f) + return; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + lreg = reg << 12; + writel(lreg, base + SSI_SACADD); + + lval = val << 4; + writel(lval , base + SSI_SACDAT); + + writel(SSI_SACNT_DEFAULT | SSI_SACNT_WR, base + SSI_SACNT); + udelay(100); +} + +static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + + unsigned short val = -1; + unsigned int lreg; + + lreg = (reg & 0x7f) << 12 ; + writel(lreg, base + SSI_SACADD); + writel(SSI_SACNT_DEFAULT | SSI_SACNT_RD, base + SSI_SACNT); + + udelay(100); + + val = (readl(base + SSI_SACDAT) >> 4) & 0xffff; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + return val; +} + +static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_reset) + imx_ssi->ac97_reset(ac97); +} + +static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_warm_reset) + imx_ssi->ac97_warm_reset(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = imx_ssi_ac97_read, + .write = imx_ssi_ac97_write, + .reset = imx_ssi_ac97_reset, + .warm_reset = imx_ssi_ac97_warm_reset +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +struct snd_soc_dai *imx_ssi_pcm_dai[2]; +EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); + +static int imx_ssi_probe(struct platform_device *pdev) +{ + struct resource *res; + struct imx_ssi *ssi; + struct imx_ssi_platform_data *pdata = pdev->dev.platform_data; + struct snd_soc_platform *platform; + int ret = 0; + unsigned int val; + + ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); + if (!ssi) + return -ENOMEM; + + if (pdata) { + ssi->ac97_reset = pdata->ac97_reset; + ssi->ac97_warm_reset = pdata->ac97_warm_reset; + ssi->flags = pdata->flags; + } + + imx_ssi_pcm_dai[pdev->id] = &ssi->dai; + + ssi->irq = platform_get_irq(pdev, 0); + + ssi->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi->clk)) { + ret = PTR_ERR(ssi->clk); + dev_err(&pdev->dev, "Cannot get the clock: %d\n", + ret); + goto failed_clk; + } + clk_enable(ssi->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + goto failed_get_resource; + } + + if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) { + dev_err(&pdev->dev, "request_mem_region failed\n"); + ret = -EBUSY; + goto failed_get_resource; + } + + ssi->base = ioremap(res->start, resource_size(res)); + if (!ssi->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENODEV; + goto failed_ioremap; + } + + if (ssi->flags & IMX_SSI_USE_AC97) { + if (ac97_ssi) { + ret = -EBUSY; + goto failed_ac97; + } + ac97_ssi = ssi; + setup_channel_to_ac97(ssi); + memcpy(&ssi->dai, &imx_ac97_dai, sizeof(imx_ac97_dai)); + } else + memcpy(&ssi->dai, &imx_ssi_dai, sizeof(imx_ssi_dai)); + + ssi->dai.id = pdev->id; + ssi->dai.dev = &pdev->dev; + ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + + writel(0x0, ssi->base + SSI_SIER); + + ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; + ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); + if (res) + ssi->dma_params_tx.dma = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); + if (res) + ssi->dma_params_rx.dma = res->start; + + ssi->dai.id = pdev->id; + ssi->dai.dev = &pdev->dev; + ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + + if ((cpu_is_mx27() || cpu_is_mx21()) && + !(ssi->flags & IMX_SSI_USE_AC97)) { + ssi->flags |= IMX_SSI_DMA; + platform = imx_ssi_dma_mx2_init(pdev, ssi); + } else + platform = imx_ssi_fiq_init(pdev, ssi); + + imx_soc_platform.pcm_ops = platform->pcm_ops; + imx_soc_platform.pcm_new = platform->pcm_new; + imx_soc_platform.pcm_free = platform->pcm_free; + + val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | + SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); + writel(val, ssi->base + SSI_SFCSR); + + ret = snd_soc_register_dai(&ssi->dai); + if (ret) { + dev_err(&pdev->dev, "register DAI failed\n"); + goto failed_register; + } + + platform_set_drvdata(pdev, ssi); + + return 0; + +failed_register: +failed_ac97: + iounmap(ssi->base); +failed_ioremap: + release_mem_region(res->start, resource_size(res)); +failed_get_resource: + clk_disable(ssi->clk); + clk_put(ssi->clk); +failed_clk: + kfree(ssi); + + return ret; +} + +static int __devexit imx_ssi_remove(struct platform_device *pdev) +{ + struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + struct imx_ssi *ssi = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&ssi->dai); + + if (ssi->flags & IMX_SSI_USE_AC97) + ac97_ssi = NULL; + + if (!(ssi->flags & IMX_SSI_DMA)) + imx_ssi_fiq_exit(pdev, ssi); + + iounmap(ssi->base); + release_mem_region(res->start, resource_size(res)); + clk_disable(ssi->clk); + clk_put(ssi->clk); + kfree(ssi); + + return 0; +} + +static struct platform_driver imx_ssi_driver = { + .probe = imx_ssi_probe, + .remove = __devexit_p(imx_ssi_remove), + + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, +}; + +static int __init imx_ssi_init(void) +{ + int ret; + + ret = snd_soc_register_platform(&imx_soc_platform); + if (ret) { + pr_err("failed to register soc platform: %d\n", ret); + return ret; + } + + ret = platform_driver_register(&imx_ssi_driver); + if (ret) { + snd_soc_unregister_platform(&imx_soc_platform); + return ret; + } + + return 0; +} + +static void __exit imx_ssi_exit(void) +{ + platform_driver_unregister(&imx_ssi_driver); + snd_soc_unregister_platform(&imx_soc_platform); +} + +module_init(imx_ssi_init); +module_exit(imx_ssi_exit); + +/* Module information */ +MODULE_AUTHOR("Sascha Hauer, "); +MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h new file mode 100644 index 0000000..cb2c81f --- /dev/null +++ b/sound/soc/imx/imx-ssi.h @@ -0,0 +1,238 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _IMX_SSI_H +#define _IMX_SSI_H + +#define SSI_STX0 0x00 +#define SSI_STX1 0x04 +#define SSI_SRX0 0x08 +#define SSI_SRX1 0x0c + +#define SSI_SCR 0x10 +#define SSI_SCR_CLK_IST (1 << 9) +#define SSI_SCR_CLK_IST_SHIFT 9 +#define SSI_SCR_TCH_EN (1 << 8) +#define SSI_SCR_SYS_CLK_EN (1 << 7) +#define SSI_SCR_I2S_MODE_NORM (0 << 5) +#define SSI_SCR_I2S_MODE_MSTR (1 << 5) +#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) +#define SSI_I2S_MODE_MASK (3 << 5) +#define SSI_SCR_SYN (1 << 4) +#define SSI_SCR_NET (1 << 3) +#define SSI_SCR_RE (1 << 2) +#define SSI_SCR_TE (1 << 1) +#define SSI_SCR_SSIEN (1 << 0) + +#define SSI_SISR 0x14 +#define SSI_SISR_MASK ((1 << 19) - 1) +#define SSI_SISR_CMDAU (1 << 18) +#define SSI_SISR_CMDDU (1 << 17) +#define SSI_SISR_RXT (1 << 16) +#define SSI_SISR_RDR1 (1 << 15) +#define SSI_SISR_RDR0 (1 << 14) +#define SSI_SISR_TDE1 (1 << 13) +#define SSI_SISR_TDE0 (1 << 12) +#define SSI_SISR_ROE1 (1 << 11) +#define SSI_SISR_ROE0 (1 << 10) +#define SSI_SISR_TUE1 (1 << 9) +#define SSI_SISR_TUE0 (1 << 8) +#define SSI_SISR_TFS (1 << 7) +#define SSI_SISR_RFS (1 << 6) +#define SSI_SISR_TLS (1 << 5) +#define SSI_SISR_RLS (1 << 4) +#define SSI_SISR_RFF1 (1 << 3) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SISR_TFE1 (1 << 1) +#define SSI_SISR_TFE0 (1 << 0) + +#define SSI_SIER 0x18 +#define SSI_SIER_RDMAE (1 << 22) +#define SSI_SIER_RIE (1 << 21) +#define SSI_SIER_TDMAE (1 << 20) +#define SSI_SIER_TIE (1 << 19) +#define SSI_SIER_CMDAU_EN (1 << 18) +#define SSI_SIER_CMDDU_EN (1 << 17) +#define SSI_SIER_RXT_EN (1 << 16) +#define SSI_SIER_RDR1_EN (1 << 15) +#define SSI_SIER_RDR0_EN (1 << 14) +#define SSI_SIER_TDE1_EN (1 << 13) +#define SSI_SIER_TDE0_EN (1 << 12) +#define SSI_SIER_ROE1_EN (1 << 11) +#define SSI_SIER_ROE0_EN (1 << 10) +#define SSI_SIER_TUE1_EN (1 << 9) +#define SSI_SIER_TUE0_EN (1 << 8) +#define SSI_SIER_TFS_EN (1 << 7) +#define SSI_SIER_RFS_EN (1 << 6) +#define SSI_SIER_TLS_EN (1 << 5) +#define SSI_SIER_RLS_EN (1 << 4) +#define SSI_SIER_RFF1_EN (1 << 3) +#define SSI_SIER_RFF0_EN (1 << 2) +#define SSI_SIER_TFE1_EN (1 << 1) +#define SSI_SIER_TFE0_EN (1 << 0) + +#define SSI_STCR 0x1c +#define SSI_STCR_TXBIT0 (1 << 9) +#define SSI_STCR_TFEN1 (1 << 8) +#define SSI_STCR_TFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_STCR_TFDIR (1 << 6) +#define SSI_STCR_TXDIR (1 << 5) +#define SSI_STCR_TSHFD (1 << 4) +#define SSI_STCR_TSCKP (1 << 3) +#define SSI_STCR_TFSI (1 << 2) +#define SSI_STCR_TFSL (1 << 1) +#define SSI_STCR_TEFS (1 << 0) + +#define SSI_SRCR 0x20 +#define SSI_SRCR_RXBIT0 (1 << 9) +#define SSI_SRCR_RFEN1 (1 << 8) +#define SSI_SRCR_RFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_SRCR_RFDIR (1 << 6) +#define SSI_SRCR_RXDIR (1 << 5) +#define SSI_SRCR_RSHFD (1 << 4) +#define SSI_SRCR_RSCKP (1 << 3) +#define SSI_SRCR_RFSI (1 << 2) +#define SSI_SRCR_RFSL (1 << 1) +#define SSI_SRCR_REFS (1 << 0) + +#define SSI_SRCCR 0x28 +#define SSI_SRCCR_DIV2 (1 << 18) +#define SSI_SRCCR_PSR (1 << 17) +#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_SRCCR_WL_MASK (0xf << 13) +#define SSI_SRCCR_DC_MASK (0x1f << 8) +#define SSI_SRCCR_PM_MASK (0xff << 0) + +#define SSI_STCCR 0x24 +#define SSI_STCCR_DIV2 (1 << 18) +#define SSI_STCCR_PSR (1 << 17) +#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_STCCR_WL_MASK (0xf << 13) +#define SSI_STCCR_DC_MASK (0x1f << 8) +#define SSI_STCCR_PM_MASK (0xff << 0) + +#define SSI_SFCSR 0x2c +#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) +#define SSI_RX_FIFO_1_COUNT_SHIFT 28 +#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) +#define SSI_TX_FIFO_1_COUNT_SHIFT 24 +#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) +#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) +#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) +#define SSI_RX_FIFO_0_COUNT_SHIFT 12 +#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) +#define SSI_TX_FIFO_0_COUNT_SHIFT 8 +#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) +#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) +#define SSI_SFCSR_RFWM0_MASK (0xf << 4) +#define SSI_SFCSR_TFWM0_MASK (0xf << 0) + +#define SSI_STR 0x30 +#define SSI_STR_TEST (1 << 15) +#define SSI_STR_RCK2TCK (1 << 14) +#define SSI_STR_RFS2TFS (1 << 13) +#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) +#define SSI_STR_TXD2RXD (1 << 7) +#define SSI_STR_TCK2RCK (1 << 6) +#define SSI_STR_TFS2RFS (1 << 5) +#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) + +#define SSI_SOR 0x34 +#define SSI_SOR_CLKOFF (1 << 6) +#define SSI_SOR_RX_CLR (1 << 5) +#define SSI_SOR_TX_CLR (1 << 4) +#define SSI_SOR_INIT (1 << 3) +#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) +#define SSI_SOR_WAIT_MASK (0x3 << 1) +#define SSI_SOR_SYNRST (1 << 0) + +#define SSI_SACNT 0x38 +#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define SSI_SACNT_WR (1 << 4) +#define SSI_SACNT_RD (1 << 3) +#define SSI_SACNT_TIF (1 << 2) +#define SSI_SACNT_FV (1 << 1) +#define SSI_SACNT_AC97EN (1 << 0) + +#define SSI_SACADD 0x3c +#define SSI_SACDAT 0x40 +#define SSI_SATAG 0x44 +#define SSI_STMSK 0x48 +#define SSI_SRMSK 0x4c +#define SSI_SACCST 0x50 +#define SSI_SACCEN 0x54 +#define SSI_SACCDIS 0x58 + +/* SSI clock sources */ +#define IMX_SSP_SYS_CLK 0 + +/* SSI audio dividers */ +#define IMX_SSI_TX_DIV_2 0 +#define IMX_SSI_TX_DIV_PSR 1 +#define IMX_SSI_TX_DIV_PM 2 +#define IMX_SSI_RX_DIV_2 3 +#define IMX_SSI_RX_DIV_PSR 4 +#define IMX_SSI_RX_DIV_PM 5 + +extern struct snd_soc_dai *imx_ssi_pcm_dai[2]; +extern struct snd_soc_platform imx_soc_platform; + +#define DRV_NAME "imx-ssi" + +struct imx_pcm_dma_params { + int dma; + unsigned long dma_addr; + int burstsize; +}; + +struct imx_ssi { + struct snd_soc_dai dai; + struct platform_device *ac97_dev; + + struct snd_soc_device imx_ac97; + struct clk *clk; + void __iomem *base; + int irq; + int fiq_enable; + unsigned int offset; + + unsigned int flags; + + void (*ac97_reset) (struct snd_ac97 *ac97); + void (*ac97_warm_reset)(struct snd_ac97 *ac97); + + struct imx_pcm_dma_params dma_params_rx; + struct imx_pcm_dma_params dma_params_tx; + + int enabled; +}; + +struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, + struct imx_ssi *ssi); +void imx_ssi_fiq_exit(struct platform_device *pdev, struct imx_ssi *ssi); +struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, + struct imx_ssi *ssi); + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm); +void imx_pcm_free(struct snd_pcm *pcm); + +/* + * Do not change this as the FIQ handler depends on this size + */ +#define IMX_SSI_DMABUF_SIZE (64 * 1024) + +#define DMA_RXFIFO_BURST 0x4 +#define DMA_TXFIFO_BURST 0x6 + +#endif /* _IMX_SSI_H */ -- cgit v0.10.2 From 157a777c8e809bd0c703e3f7617b3539df30feff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Jan 2010 17:50:29 +0000 Subject: ASoC: Fix i.MX audio build for i.MX3x Don't unconditionally include the i.MX2x DMA driver, the arch/arm functions it uses aren't available for i.MX3x. Signed-off-by: Mark Brown Acked-by: Javier Martin diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 4bde34a..d05cc95 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,5 +1,5 @@ # i.MX Platform Support -snd-soc-imx-objs := imx-ssi.o imx-pcm-fiq.o imx-pcm-dma-mx2.o +snd-soc-imx-objs := imx-ssi.o imx-pcm-fiq.o ifdef CONFIG_MACH_MX27 snd-soc-imx-objs += imx-pcm-dma-mx2.o -- cgit v0.10.2 From 48dbc41988d07c7a9ba83afd31543d8ecb2beecc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Jan 2010 17:56:52 +0000 Subject: ASoC: Convert new i.MX SSI driver to use static DAI array While dynamically allocated DAIs are the way forward the core doesn't yet support anything except matching with a pointer to the actual DAI so convert to doing that so that machine drivers don't have to jump through hoops to register themselves. Signed-off-by: Mark Brown Acked-by: Javier Martin diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index c57a11f..ccb7ec9 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -60,7 +60,7 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 sccr; sccr = readl(ssi->base + SSI_STCCR); @@ -87,7 +87,7 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, */ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 strcr = 0, scr; scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET); @@ -160,7 +160,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 scr; scr = readl(ssi->base + SSI_SCR); @@ -188,7 +188,7 @@ static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 stccr, srccr; stccr = readl(ssi->base + SSI_STCCR); @@ -237,7 +237,7 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; u32 reg, sccr; /* Tx/Rx config */ @@ -274,7 +274,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + struct imx_ssi *ssi = cpu_dai->private_data; unsigned int sier_bits, sier; unsigned int scr; @@ -570,7 +570,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = { }; EXPORT_SYMBOL_GPL(soc_ac97_ops); -struct snd_soc_dai *imx_ssi_pcm_dai[2]; +struct snd_soc_dai imx_ssi_pcm_dai[2]; EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); static int imx_ssi_probe(struct platform_device *pdev) @@ -581,6 +581,10 @@ static int imx_ssi_probe(struct platform_device *pdev) struct snd_soc_platform *platform; int ret = 0; unsigned int val; + struct snd_soc_dai *dai = &imx_ssi_pcm_dai[pdev->id]; + + if (dai->id >= ARRAY_SIZE(imx_ssi_pcm_dai)) + return -EINVAL; ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); if (!ssi) @@ -592,8 +596,6 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->flags = pdata->flags; } - imx_ssi_pcm_dai[pdev->id] = &ssi->dai; - ssi->irq = platform_get_irq(pdev, 0); ssi->clk = clk_get(&pdev->dev, NULL); @@ -631,13 +633,9 @@ static int imx_ssi_probe(struct platform_device *pdev) } ac97_ssi = ssi; setup_channel_to_ac97(ssi); - memcpy(&ssi->dai, &imx_ac97_dai, sizeof(imx_ac97_dai)); + memcpy(dai, &imx_ac97_dai, sizeof(imx_ac97_dai)); } else - memcpy(&ssi->dai, &imx_ssi_dai, sizeof(imx_ssi_dai)); - - ssi->dai.id = pdev->id; - ssi->dai.dev = &pdev->dev; - ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + memcpy(dai, &imx_ssi_dai, sizeof(imx_ssi_dai)); writel(0x0, ssi->base + SSI_SIER); @@ -652,9 +650,10 @@ static int imx_ssi_probe(struct platform_device *pdev) if (res) ssi->dma_params_rx.dma = res->start; - ssi->dai.id = pdev->id; - ssi->dai.dev = &pdev->dev; - ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + dai->id = pdev->id; + dai->dev = &pdev->dev; + dai->name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + dai->private_data = ssi; if ((cpu_is_mx27() || cpu_is_mx21()) && !(ssi->flags & IMX_SSI_USE_AC97)) { @@ -671,7 +670,7 @@ static int imx_ssi_probe(struct platform_device *pdev) SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); writel(val, ssi->base + SSI_SFCSR); - ret = snd_soc_register_dai(&ssi->dai); + ret = snd_soc_register_dai(dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); goto failed_register; @@ -699,8 +698,9 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev) { struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); + struct snd_soc_dai *dai = &imx_ssi_pcm_dai[pdev->id]; - snd_soc_unregister_dai(&ssi->dai); + snd_soc_unregister_dai(dai); if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index cb2c81f..55f26eb 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -183,7 +183,7 @@ #define IMX_SSI_RX_DIV_PSR 4 #define IMX_SSI_RX_DIV_PM 5 -extern struct snd_soc_dai *imx_ssi_pcm_dai[2]; +extern struct snd_soc_dai imx_ssi_pcm_dai[2]; extern struct snd_soc_platform imx_soc_platform; #define DRV_NAME "imx-ssi" @@ -195,7 +195,6 @@ struct imx_pcm_dma_params { }; struct imx_ssi { - struct snd_soc_dai dai; struct platform_device *ac97_dev; struct snd_soc_device imx_ac97; -- cgit v0.10.2 From d08a68bfca5a6464eb9167be0659bf0676f02555 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jan 2010 16:56:19 +0000 Subject: ASoC: i.MX SSI driver does not yet support master mode The clocks for the SSI block need handling before this can work. Signed-off-by: Mark Brown diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index ccb7ec9..56f46a7 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -133,15 +133,11 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* DAI clock master masks */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - strcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; - break; - case SND_SOC_DAIFMT_CBM_CFS: - strcr |= SSI_STCR_TFDIR; - break; - case SND_SOC_DAIFMT_CBS_CFM: - strcr |= SSI_STCR_TXDIR; + case SND_SOC_DAIFMT_CBM_CFM: break; + default: + /* Master mode not implemented, needs handling of clocks. */ + return -EINVAL; } strcr |= SSI_STCR_TFEN0; -- cgit v0.10.2 From e919c24b6422a095bed3929074bd74ae1dbf251f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 17 Jan 2010 11:08:38 +0000 Subject: ASoC: Remove old i.MX driver code This has been superceeded by Sascha's new driver but was not removed in the patch series due to cutdowns for review. Signed-off-by: Mark Brown diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c deleted file mode 100644 index bffffcd..0000000 --- a/sound/soc/imx/mx1_mx2-pcm.c +++ /dev/null @@ -1,488 +0,0 @@ -/* - * mx1_mx2-pcm.c -- ALSA SoC interface for Freescale i.MX1x, i.MX2x CPUs - * - * Copyright 2009 Vista Silicon S.L. - * Author: Javier Martin - * javier.martin@vista-silicon.com - * - * Based on mxc-pcm.c by Liam Girdwood. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "mx1_mx2-pcm.h" - - -static const struct snd_pcm_hardware mx1_mx2_pcm_hardware = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .buffer_bytes_max = 32 * 1024, - .period_bytes_min = 64, - .period_bytes_max = 8 * 1024, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -struct mx1_mx2_runtime_data { - int dma_ch; - int active; - unsigned int period; - unsigned int periods; - int tx_spin; - spinlock_t dma_lock; - struct mx1_mx2_pcm_dma_params *dma_params; -}; - - -/** - * This function stops the current dma transfer for playback - * and clears the dma pointers. - * - * @param substream pointer to the structure of the current stream. - * - */ -static int audio_stop_dma(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - unsigned long flags; - - spin_lock_irqsave(&prtd->dma_lock, flags); - - pr_debug("%s\n", __func__); - - prtd->active = 0; - prtd->period = 0; - prtd->periods = 0; - - /* this stops the dma channel and clears the buffer ptrs */ - - imx_dma_disable(prtd->dma_ch); - - spin_unlock_irqrestore(&prtd->dma_lock, flags); - - return 0; -} - -/** - * This function is called whenever a new audio block needs to be - * transferred to the codec. The function receives the address and the size - * of the new block and start a new DMA transfer. - * - * @param substream pointer to the structure of the current stream. - * - */ -static int dma_new_period(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - unsigned int dma_size; - unsigned int offset; - int ret = 0; - dma_addr_t mem_addr; - unsigned int dev_addr; - - if (prtd->active) { - dma_size = frames_to_bytes(runtime, runtime->period_size); - offset = dma_size * prtd->period; - - pr_debug("%s: period (%d) out of (%d)\n", __func__, - prtd->period, - runtime->periods); - pr_debug("period_size %d frames\n offset %d bytes\n", - (unsigned int)runtime->period_size, - offset); - pr_debug("dma_size %d bytes\n", dma_size); - - snd_BUG_ON(dma_size > mx1_mx2_pcm_hardware.period_bytes_max); - - mem_addr = (dma_addr_t)(runtime->dma_addr + offset); - dev_addr = prtd->dma_params->per_address; - pr_debug("%s: mem_addr is %x\n dev_addr is %x\n", - __func__, mem_addr, dev_addr); - - ret = imx_dma_setup_single(prtd->dma_ch, mem_addr, - dma_size, dev_addr, - prtd->dma_params->transfer_type); - if (ret < 0) { - printk(KERN_ERR "Error %d configuring DMA\n", ret); - return ret; - } - imx_dma_enable(prtd->dma_ch); - - pr_debug("%s: transfer enabled\nmem_addr = %x\n", - __func__, (unsigned int) mem_addr); - pr_debug("dev_addr = %x\ndma_size = %d\n", - (unsigned int) dev_addr, dma_size); - - prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ - prtd->period++; - prtd->period %= runtime->periods; - } - return ret; -} - - -/** - * This is a callback which will be called - * when a TX transfer finishes. The call occurs - * in interrupt context. - * - * @param dat pointer to the structure of the current stream. - * - */ -static void audio_dma_irq(int channel, void *data) -{ - struct snd_pcm_substream *substream; - struct snd_pcm_runtime *runtime; - struct mx1_mx2_runtime_data *prtd; - unsigned int dma_size; - unsigned int previous_period; - unsigned int offset; - - substream = data; - runtime = substream->runtime; - prtd = runtime->private_data; - previous_period = prtd->periods; - dma_size = frames_to_bytes(runtime, runtime->period_size); - offset = dma_size * previous_period; - - prtd->tx_spin = 0; - prtd->periods++; - prtd->periods %= runtime->periods; - - pr_debug("%s: irq per %d offset %x\n", __func__, prtd->periods, offset); - - /* - * If we are getting a callback for an active stream then we inform - * the PCM middle layer we've finished a period - */ - if (prtd->active) - snd_pcm_period_elapsed(substream); - - /* - * Trig next DMA transfer - */ - dma_new_period(substream); -} - -/** - * This function configures the hardware to allow audio - * playback operations. It is called by ALSA framework. - * - * @param substream pointer to the structure of the current stream. - * - * @return 0 on success, -1 otherwise. - */ -static int -snd_mx1_mx2_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - - prtd->period = 0; - prtd->periods = 0; - - return 0; -} - -static int mx1_mx2_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - if (ret < 0) { - printk(KERN_ERR "%s: Error %d failed to malloc pcm pages \n", - __func__, ret); - return ret; - } - - pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_addr 0x(%x)\n", - __func__, (unsigned int)runtime->dma_addr); - pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_area 0x(%x)\n", - __func__, (unsigned int)runtime->dma_area); - pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_bytes 0x(%x)\n", - __func__, (unsigned int)runtime->dma_bytes); - - return ret; -} - -static int mx1_mx2_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - - imx_dma_free(prtd->dma_ch); - - snd_pcm_lib_free_pages(substream); - - return 0; -} - -static int mx1_mx2_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct mx1_mx2_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->tx_spin = 0; - /* requested stream startup */ - prtd->active = 1; - pr_debug("%s: starting dma_new_period\n", __func__); - ret = dma_new_period(substream); - break; - case SNDRV_PCM_TRIGGER_STOP: - /* requested stream shutdown */ - pr_debug("%s: stopping dma transfer\n", __func__); - ret = audio_stop_dma(substream); - break; - default: - ret = -EINVAL; - break; - } - - return ret; -} - -static snd_pcm_uframes_t -mx1_mx2_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - unsigned int offset = 0; - - /* tx_spin value is used here to check if a transfer is active */ - if (prtd->tx_spin) { - offset = (runtime->period_size * (prtd->periods)) + - (runtime->period_size >> 1); - if (offset >= runtime->buffer_size) - offset = runtime->period_size >> 1; - } else { - offset = (runtime->period_size * (prtd->periods)); - if (offset >= runtime->buffer_size) - offset = 0; - } - pr_debug("%s: pointer offset %x\n", __func__, offset); - - return offset; -} - -static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mx1_mx2_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; - int ret; - - snd_soc_set_runtime_hwparams(substream, &mx1_mx2_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - prtd = kzalloc(sizeof(struct mx1_mx2_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - - runtime->private_data = prtd; - - if (!dma_data) - return -ENODEV; - - prtd->dma_params = dma_data; - - pr_debug("%s: Requesting dma channel (%s)\n", __func__, - prtd->dma_params->name); - ret = imx_dma_request_by_prio(prtd->dma_params->name, DMA_PRIO_HIGH); - if (ret < 0) { - printk(KERN_ERR "Error %d requesting dma channel\n", ret); - return ret; - } - prtd->dma_ch = ret; - imx_dma_config_burstlen(prtd->dma_ch, - prtd->dma_params->watermark_level); - - ret = imx_dma_config_channel(prtd->dma_ch, - prtd->dma_params->per_config, - prtd->dma_params->mem_config, - prtd->dma_params->event_id, 0); - - if (ret) { - pr_debug(KERN_ERR "Error %d configuring dma channel %d\n", - ret, prtd->dma_ch); - return ret; - } - - pr_debug("%s: Setting tx dma callback function\n", __func__); - ret = imx_dma_setup_handlers(prtd->dma_ch, - audio_dma_irq, NULL, - (void *)substream); - if (ret < 0) { - printk(KERN_ERR "Error %d setting dma callback function\n", ret); - return ret; - } - return 0; - - out: - return ret; -} - -static int mx1_mx2_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mx1_mx2_runtime_data *prtd = runtime->private_data; - - kfree(prtd); - - return 0; -} - -static int mx1_mx2_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops mx1_mx2_pcm_ops = { - .open = mx1_mx2_pcm_open, - .close = mx1_mx2_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = mx1_mx2_pcm_hw_params, - .hw_free = mx1_mx2_pcm_hw_free, - .prepare = snd_mx1_mx2_prepare, - .trigger = mx1_mx2_pcm_trigger, - .pointer = mx1_mx2_pcm_pointer, - .mmap = mx1_mx2_pcm_mmap, -}; - -static u64 mx1_mx2_pcm_dmamask = 0xffffffff; - -static int mx1_mx2_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = mx1_mx2_pcm_hardware.buffer_bytes_max; - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - - /* Reserve uncached-buffered memory area for DMA */ - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - pr_debug("%s: preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - __func__, (void *) buf->area, (void *) buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static void mx1_mx2_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static int mx1_mx2_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &mx1_mx2_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - pr_debug("%s: preallocate playback buffer\n", __func__); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - pr_debug("%s: preallocate capture buffer\n", __func__); - if (ret) - goto out; - } - out: - return ret; -} - -struct snd_soc_platform mx1_mx2_soc_platform = { - .name = "mx1_mx2-audio", - .pcm_ops = &mx1_mx2_pcm_ops, - .pcm_new = mx1_mx2_pcm_new, - .pcm_free = mx1_mx2_pcm_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(mx1_mx2_soc_platform); - -static int __init mx1_mx2_soc_platform_init(void) -{ - return snd_soc_register_platform(&mx1_mx2_soc_platform); -} -module_init(mx1_mx2_soc_platform_init); - -static void __exit mx1_mx2_soc_platform_exit(void) -{ - snd_soc_unregister_platform(&mx1_mx2_soc_platform); -} -module_exit(mx1_mx2_soc_platform_exit); - -MODULE_AUTHOR("Javier Martin, javier.martin@vista-silicon.com"); -MODULE_DESCRIPTION("Freescale i.MX2x, i.MX1x PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mx1_mx2-pcm.h b/sound/soc/imx/mx1_mx2-pcm.h deleted file mode 100644 index 2e52810..0000000 --- a/sound/soc/imx/mx1_mx2-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * mx1_mx2-pcm.h :- ASoC platform header for Freescale i.MX1x, i.MX2x - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _MX1_MX2_PCM_H -#define _MX1_MX2_PCM_H - -/* DMA information for mx1_mx2 platforms */ -struct mx1_mx2_pcm_dma_params { - char *name; /* stream identifier */ - unsigned int transfer_type; /* READ or WRITE DMA transfer */ - dma_addr_t per_address; /* physical address of SSI fifo */ - int event_id; /* fixed DMA number for SSI fifo */ - int watermark_level; /* SSI fifo watermark level */ - int per_config; /* DMA Config flags for peripheral */ - int mem_config; /* DMA Config flags for RAM */ - }; - -/* platform data */ -extern struct snd_soc_platform mx1_mx2_soc_platform; - -#endif diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c deleted file mode 100644 index 0267d2d..0000000 --- a/sound/soc/imx/mx27vis_wm8974.c +++ /dev/null @@ -1,317 +0,0 @@ -/* - * mx27vis_wm8974.c -- SoC audio for mx27vis - * - * Copyright 2009 Vista Silicon S.L. - * Author: Javier Martin - * javier.martin@vista-silicon.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include - - -#include "../codecs/wm8974.h" -#include "mx1_mx2-pcm.h" -#include "mxc-ssi.h" -#include -#include - -#define IGNORED_ARG 0 - - -static struct snd_soc_card mx27vis; - -/** - * This function connects SSI1 (HPCR1) as slave to - * SSI1 external signals (PPCR1) - * As slave, HPCR1 must set TFSDIR and TCLKDIR as inputs from - * port 4 - */ -void audmux_connect_1_4(void) -{ - pr_debug("AUDMUX: normal operation mode\n"); - /* Reset HPCR1 and PPCR1 */ - - DAM_HPCR1 = 0x00000000; - DAM_PPCR1 = 0x00000000; - - /* set to synchronous */ - DAM_HPCR1 |= AUDMUX_HPCR_SYN; - DAM_PPCR1 |= AUDMUX_PPCR_SYN; - - - /* set Rx sources 1 <--> 4 */ - DAM_HPCR1 |= AUDMUX_HPCR_RXDSEL(3); /* port 4 */ - DAM_PPCR1 |= AUDMUX_PPCR_RXDSEL(0); /* port 1 */ - - /* set Tx frame and Clock direction and source 4 --> 1 output */ - DAM_HPCR1 |= AUDMUX_HPCR_TFSDIR | AUDMUX_HPCR_TCLKDIR; - DAM_HPCR1 |= AUDMUX_HPCR_TFCSEL(3); /* TxDS and TxCclk from port 4 */ - - return; -} - -static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int pll_out = 0, bclk = 0, fmt = 0, mclk = 0; - int ret = 0; - - /* - * The WM8974 is better at generating accurate audio clocks than the - * MX27 SSI controller, so we will use it as master when we can. - */ - switch (params_rate(params)) { - case 8000: - fmt = SND_SOC_DAIFMT_CBM_CFM; - mclk = WM8974_MCLKDIV_12; - pll_out = 24576000; - break; - case 16000: - fmt = SND_SOC_DAIFMT_CBM_CFM; - pll_out = 12288000; - break; - case 48000: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_4; - pll_out = 12288000; - break; - case 96000: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_2; - pll_out = 12288000; - break; - case 11025: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_16; - pll_out = 11289600; - break; - case 22050: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_8; - pll_out = 11289600; - break; - case 44100: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_4; - mclk = WM8974_MCLKDIV_2; - pll_out = 11289600; - break; - case 88200: - fmt = SND_SOC_DAIFMT_CBM_CFM; - bclk = WM8974_BCLKDIV_2; - pll_out = 11289600; - break; - } - - /* set codec DAI configuration */ - ret = codec_dai->ops->set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_SYNC | fmt); - if (ret < 0) { - printk(KERN_ERR "Error from codec DAI configuration\n"); - return ret; - } - - /* set cpu DAI configuration */ - ret = cpu_dai->ops->set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_SYNC | fmt); - if (ret < 0) { - printk(KERN_ERR "Error from cpu DAI configuration\n"); - return ret; - } - - /* Put DC field of STCCR to 1 (not zero) */ - ret = cpu_dai->ops->set_tdm_slot(cpu_dai, 0, 2); - - /* set the SSI system clock as input */ - ret = cpu_dai->ops->set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "Error when setting system SSI clk\n"); - return ret; - } - - /* set codec BCLK division for sample rate */ - ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_BCLKDIV, bclk); - if (ret < 0) { - printk(KERN_ERR "Error when setting BCLK division\n"); - return ret; - } - - - /* codec PLL input is 25 MHz */ - ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG, - 25000000, pll_out); - if (ret < 0) { - printk(KERN_ERR "Error when setting PLL input\n"); - return ret; - } - - /*set codec MCLK division for sample rate */ - ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_MCLKDIV, mclk); - if (ret < 0) { - printk(KERN_ERR "Error when setting MCLK division\n"); - return ret; - } - - return 0; -} - -static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - /* disable the PLL */ - return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); -} - -/* - * mx27vis WM8974 HiFi DAI opserations. - */ -static struct snd_soc_ops mx27vis_hifi_ops = { - .hw_params = mx27vis_hifi_hw_params, - .hw_free = mx27vis_hifi_hw_free, -}; - - -static int mx27vis_suspend(struct platform_device *pdev, pm_message_t state) -{ - return 0; -} - -static int mx27vis_resume(struct platform_device *pdev) -{ - return 0; -} - -static int mx27vis_probe(struct platform_device *pdev) -{ - int ret = 0; - - ret = get_ssi_clk(0, &pdev->dev); - - if (ret < 0) { - printk(KERN_ERR "%s: cant get ssi clock\n", __func__); - return ret; - } - - - return 0; -} - -static int mx27vis_remove(struct platform_device *pdev) -{ - put_ssi_clk(0); - return 0; -} - -static struct snd_soc_dai_link mx27vis_dai[] = { -{ /* Hifi Playback*/ - .name = "WM8974", - .stream_name = "WM8974 HiFi", - .cpu_dai = &imx_ssi_pcm_dai[0], - .codec_dai = &wm8974_dai, - .ops = &mx27vis_hifi_ops, -}, -}; - -static struct snd_soc_card mx27vis = { - .name = "mx27vis", - .platform = &mx1_mx2_soc_platform, - .probe = mx27vis_probe, - .remove = mx27vis_remove, - .suspend_pre = mx27vis_suspend, - .resume_post = mx27vis_resume, - .dai_link = mx27vis_dai, - .num_links = ARRAY_SIZE(mx27vis_dai), -}; - -static struct snd_soc_device mx27vis_snd_devdata = { - .card = &mx27vis, - .codec_dev = &soc_codec_dev_wm8974, -}; - -static struct platform_device *mx27vis_snd_device; - -/* Temporal definition of board specific behaviour */ -void gpio_ssi_active(int ssi_num) -{ - int ret = 0; - - unsigned int ssi1_pins[] = { - PC20_PF_SSI1_FS, - PC21_PF_SSI1_RXD, - PC22_PF_SSI1_TXD, - PC23_PF_SSI1_CLK, - }; - unsigned int ssi2_pins[] = { - PC24_PF_SSI2_FS, - PC25_PF_SSI2_RXD, - PC26_PF_SSI2_TXD, - PC27_PF_SSI2_CLK, - }; - if (ssi_num == 0) - ret = mxc_gpio_setup_multiple_pins(ssi1_pins, - ARRAY_SIZE(ssi1_pins), "USB OTG"); - else - ret = mxc_gpio_setup_multiple_pins(ssi2_pins, - ARRAY_SIZE(ssi2_pins), "USB OTG"); - if (ret) - printk(KERN_ERR "Error requesting ssi %x pins\n", ssi_num); -} - - -static int __init mx27vis_init(void) -{ - int ret; - - mx27vis_snd_device = platform_device_alloc("soc-audio", -1); - if (!mx27vis_snd_device) - return -ENOMEM; - - platform_set_drvdata(mx27vis_snd_device, &mx27vis_snd_devdata); - mx27vis_snd_devdata.dev = &mx27vis_snd_device->dev; - ret = platform_device_add(mx27vis_snd_device); - - if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(mx27vis_snd_device); - } - - /* WM8974 uses SSI1 (HPCR1) via AUDMUX port 4 for audio (PPCR1) */ - gpio_ssi_active(0); - audmux_connect_1_4(); - - return ret; -} - -static void __exit mx27vis_exit(void) -{ - /* We should call some "ssi_gpio_inactive()" properly */ -} - -module_init(mx27vis_init); -module_exit(mx27vis_exit); - - -MODULE_AUTHOR("Javier Martin, javier.martin@vista-silicon.com"); -MODULE_DESCRIPTION("ALSA SoC WM8974 mx27vis"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c deleted file mode 100644 index ccdefe6..0000000 --- a/sound/soc/imx/mxc-ssi.c +++ /dev/null @@ -1,860 +0,0 @@ -/* - * mxc-ssi.c -- SSI driver for Freescale IMX - * - * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Based on mxc-alsa-mc13783 (C) 2006 Freescale. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * TODO: - * Need to rework SSI register defs when new defs go into mainline. - * Add support for TDM and FIFO 1. - * Add support for i.mx3x DMA interface. - * - */ - - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "mxc-ssi.h" -#include "mx1_mx2-pcm.h" - -#define SSI1_PORT 0 -#define SSI2_PORT 1 - -static int ssi_active[2] = {0, 0}; - -/* DMA information for mx1_mx2 platforms */ -static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out0 = { - .name = "SSI1 PCM Stereo out 0", - .transfer_type = DMA_MODE_WRITE, - .per_address = SSI1_BASE_ADDR + STX0, - .event_id = DMA_REQ_SSI1_TX0, - .watermark_level = TXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out1 = { - .name = "SSI1 PCM Stereo out 1", - .transfer_type = DMA_MODE_WRITE, - .per_address = SSI1_BASE_ADDR + STX1, - .event_id = DMA_REQ_SSI1_TX1, - .watermark_level = TXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in0 = { - .name = "SSI1 PCM Stereo in 0", - .transfer_type = DMA_MODE_READ, - .per_address = SSI1_BASE_ADDR + SRX0, - .event_id = DMA_REQ_SSI1_RX0, - .watermark_level = RXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in1 = { - .name = "SSI1 PCM Stereo in 1", - .transfer_type = DMA_MODE_READ, - .per_address = SSI1_BASE_ADDR + SRX1, - .event_id = DMA_REQ_SSI1_RX1, - .watermark_level = RXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out0 = { - .name = "SSI2 PCM Stereo out 0", - .transfer_type = DMA_MODE_WRITE, - .per_address = SSI2_BASE_ADDR + STX0, - .event_id = DMA_REQ_SSI2_TX0, - .watermark_level = TXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out1 = { - .name = "SSI2 PCM Stereo out 1", - .transfer_type = DMA_MODE_WRITE, - .per_address = SSI2_BASE_ADDR + STX1, - .event_id = DMA_REQ_SSI2_TX1, - .watermark_level = TXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in0 = { - .name = "SSI2 PCM Stereo in 0", - .transfer_type = DMA_MODE_READ, - .per_address = SSI2_BASE_ADDR + SRX0, - .event_id = DMA_REQ_SSI2_RX0, - .watermark_level = RXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in1 = { - .name = "SSI2 PCM Stereo in 1", - .transfer_type = DMA_MODE_READ, - .per_address = SSI2_BASE_ADDR + SRX1, - .event_id = DMA_REQ_SSI2_RX1, - .watermark_level = RXFIFO_WATERMARK, - .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, -}; - -static struct clk *ssi_clk0, *ssi_clk1; - -int get_ssi_clk(int ssi, struct device *dev) -{ - switch (ssi) { - case 0: - ssi_clk0 = clk_get(dev, "ssi1"); - if (IS_ERR(ssi_clk0)) - return PTR_ERR(ssi_clk0); - return 0; - case 1: - ssi_clk1 = clk_get(dev, "ssi2"); - if (IS_ERR(ssi_clk1)) - return PTR_ERR(ssi_clk1); - return 0; - default: - return -EINVAL; - } -} -EXPORT_SYMBOL(get_ssi_clk); - -void put_ssi_clk(int ssi) -{ - switch (ssi) { - case 0: - clk_put(ssi_clk0); - ssi_clk0 = NULL; - break; - case 1: - clk_put(ssi_clk1); - ssi_clk1 = NULL; - break; - } -} -EXPORT_SYMBOL(put_ssi_clk); - -/* - * SSI system clock configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - u32 scr; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - scr = SSI1_SCR; - pr_debug("%s: SCR for SSI1 is %x\n", __func__, scr); - } else { - scr = SSI2_SCR; - pr_debug("%s: SCR for SSI2 is %x\n", __func__, scr); - } - - if (scr & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - - switch (clk_id) { - case IMX_SSP_SYS_CLK: - if (dir == SND_SOC_CLOCK_OUT) { - scr |= SSI_SCR_SYS_CLK_EN; - pr_debug("%s: clk of is output\n", __func__); - } else { - scr &= ~SSI_SCR_SYS_CLK_EN; - pr_debug("%s: clk of is input\n", __func__); - } - break; - default: - return -EINVAL; - } - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - pr_debug("%s: writeback of SSI1_SCR\n", __func__); - SSI1_SCR = scr; - } else { - pr_debug("%s: writeback of SSI2_SCR\n", __func__); - SSI2_SCR = scr; - } - - return 0; -} - -/* - * SSI Clock dividers - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - u32 stccr, srccr; - - pr_debug("%s\n", __func__); - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - if (SSI1_SCR & SSI_SCR_SSIEN) - return 0; - srccr = SSI1_STCCR; - stccr = SSI1_STCCR; - } else { - if (SSI2_SCR & SSI_SCR_SSIEN) - return 0; - srccr = SSI2_STCCR; - stccr = SSI2_STCCR; - } - - switch (div_id) { - case IMX_SSI_TX_DIV_2: - stccr &= ~SSI_STCCR_DIV2; - stccr |= div; - break; - case IMX_SSI_TX_DIV_PSR: - stccr &= ~SSI_STCCR_PSR; - stccr |= div; - break; - case IMX_SSI_TX_DIV_PM: - stccr &= ~0xff; - stccr |= SSI_STCCR_PM(div); - break; - case IMX_SSI_RX_DIV_2: - stccr &= ~SSI_STCCR_DIV2; - stccr |= div; - break; - case IMX_SSI_RX_DIV_PSR: - stccr &= ~SSI_STCCR_PSR; - stccr |= div; - break; - case IMX_SSI_RX_DIV_PM: - stccr &= ~0xff; - stccr |= SSI_STCCR_PM(div); - break; - default: - return -EINVAL; - } - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_STCCR = stccr; - SSI1_SRCCR = srccr; - } else { - SSI2_STCCR = stccr; - SSI2_SRCCR = srccr; - } - return 0; -} - -/* - * SSI Network Mode or TDM slots configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) -{ - u32 stmsk, srmsk, stccr; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - if (SSI1_SCR & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - stccr = SSI1_STCCR; - } else { - if (SSI2_SCR & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - stccr = SSI2_STCCR; - } - - stmsk = srmsk = mask; - stccr &= ~SSI_STCCR_DC_MASK; - stccr |= SSI_STCCR_DC(slots - 1); - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_STMSK = stmsk; - SSI1_SRMSK = srmsk; - SSI1_SRCCR = SSI1_STCCR = stccr; - } else { - SSI2_STMSK = stmsk; - SSI2_SRMSK = srmsk; - SSI2_SRCCR = SSI2_STCCR = stccr; - } - - return 0; -} - -/* - * SSI DAI format configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - * Note: We don't use the I2S modes but instead manually configure the - * SSI for I2S. - */ -static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - u32 stcr = 0, srcr = 0, scr; - - /* - * This is done to avoid this function to modify - * previous set values in stcr - */ - stcr = SSI1_STCR; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - scr = SSI1_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); - else - scr = SSI2_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); - - if (scr & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - - /* DAI mode */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - /* data on rising edge of bclk, frame low 1clk before data */ - stcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; - srcr |= SSI_SRCR_RFSI | SSI_SRCR_REFS | SSI_SRCR_RXBIT0; - break; - case SND_SOC_DAIFMT_LEFT_J: - /* data on rising edge of bclk, frame high with data */ - stcr |= SSI_STCR_TXBIT0; - srcr |= SSI_SRCR_RXBIT0; - break; - case SND_SOC_DAIFMT_DSP_B: - /* data on rising edge of bclk, frame high with data */ - stcr |= SSI_STCR_TFSL; - srcr |= SSI_SRCR_RFSL; - break; - case SND_SOC_DAIFMT_DSP_A: - /* data on rising edge of bclk, frame high 1clk before data */ - stcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; - srcr |= SSI_SRCR_RFSL | SSI_SRCR_REFS; - break; - } - - /* DAI clock inversion */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_IF: - stcr |= SSI_STCR_TFSI; - stcr &= ~SSI_STCR_TSCKP; - srcr |= SSI_SRCR_RFSI; - srcr &= ~SSI_SRCR_RSCKP; - break; - case SND_SOC_DAIFMT_IB_NF: - stcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); - srcr &= ~(SSI_SRCR_RSCKP | SSI_SRCR_RFSI); - break; - case SND_SOC_DAIFMT_NB_IF: - stcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; - srcr |= SSI_SRCR_RFSI | SSI_SRCR_RSCKP; - break; - case SND_SOC_DAIFMT_NB_NF: - stcr &= ~SSI_STCR_TFSI; - stcr |= SSI_STCR_TSCKP; - srcr &= ~SSI_SRCR_RFSI; - srcr |= SSI_SRCR_RSCKP; - break; - } - - /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - stcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; - srcr |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; - break; - case SND_SOC_DAIFMT_CBM_CFS: - stcr |= SSI_STCR_TFDIR; - srcr |= SSI_SRCR_RFDIR; - break; - case SND_SOC_DAIFMT_CBS_CFM: - stcr |= SSI_STCR_TXDIR; - srcr |= SSI_SRCR_RXDIR; - break; - } - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_STCR = stcr; - SSI1_SRCR = srcr; - SSI1_SCR = scr; - } else { - SSI2_STCR = stcr; - SSI2_SRCR = srcr; - SSI2_SCR = scr; - } - - return 0; -} - -static int imx_ssi_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set up TX DMA params */ - switch (cpu_dai->id) { - case IMX_DAI_SSI0: - cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out0; - break; - case IMX_DAI_SSI1: - cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out1; - break; - case IMX_DAI_SSI2: - cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out0; - break; - case IMX_DAI_SSI3: - cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out1; - } - pr_debug("%s: (playback)\n", __func__); - } else { - /* set up RX DMA params */ - switch (cpu_dai->id) { - case IMX_DAI_SSI0: - cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in0; - break; - case IMX_DAI_SSI1: - cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in1; - break; - case IMX_DAI_SSI2: - cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in0; - break; - case IMX_DAI_SSI3: - cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in1; - } - pr_debug("%s: (capture)\n", __func__); - } - - /* - * we cant really change any SSI values after SSI is enabled - * need to fix in software for max flexibility - lrg - */ - if (cpu_dai->active) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - - /* reset the SSI port - Sect 45.4.4 */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - - if (!ssi_clk0) - return -EINVAL; - - if (ssi_active[SSI1_PORT]++) { - pr_debug("%s: exit before reset\n", __func__); - return 0; - } - - /* SSI1 Reset */ - SSI1_SCR = 0; - - SSI1_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | - SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | - SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | - SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); - } else { - - if (!ssi_clk1) - return -EINVAL; - - if (ssi_active[SSI2_PORT]++) { - pr_debug("%s: exit before reset\n", __func__); - return 0; - } - - /* SSI2 Reset */ - SSI2_SCR = 0; - - SSI2_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | - SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | - SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | - SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); - } - - return 0; -} - -int imx_ssi_hw_tx_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - u32 stccr, stcr, sier; - - pr_debug("%s\n", __func__); - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - stccr = SSI1_STCCR & ~SSI_STCCR_WL_MASK; - stcr = SSI1_STCR; - sier = SSI1_SIER; - } else { - stccr = SSI2_STCCR & ~SSI_STCCR_WL_MASK; - stcr = SSI2_STCR; - sier = SSI2_SIER; - } - - /* DAI data (word) size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - stccr |= SSI_STCCR_WL(16); - break; - case SNDRV_PCM_FORMAT_S20_3LE: - stccr |= SSI_STCCR_WL(20); - break; - case SNDRV_PCM_FORMAT_S24_LE: - stccr |= SSI_STCCR_WL(24); - break; - } - - /* enable interrupts */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - stcr |= SSI_STCR_TFEN0; - else - stcr |= SSI_STCR_TFEN1; - sier |= SSI_SIER_TDMAE; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_STCR = stcr; - SSI1_STCCR = stccr; - SSI1_SIER = sier; - } else { - SSI2_STCR = stcr; - SSI2_STCCR = stccr; - SSI2_SIER = sier; - } - - return 0; -} - -int imx_ssi_hw_rx_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - u32 srccr, srcr, sier; - - pr_debug("%s\n", __func__); - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - srccr = SSI1_SRCCR & ~SSI_SRCCR_WL_MASK; - srcr = SSI1_SRCR; - sier = SSI1_SIER; - } else { - srccr = SSI2_SRCCR & ~SSI_SRCCR_WL_MASK; - srcr = SSI2_SRCR; - sier = SSI2_SIER; - } - - /* DAI data (word) size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - srccr |= SSI_SRCCR_WL(16); - break; - case SNDRV_PCM_FORMAT_S20_3LE: - srccr |= SSI_SRCCR_WL(20); - break; - case SNDRV_PCM_FORMAT_S24_LE: - srccr |= SSI_SRCCR_WL(24); - break; - } - - /* enable interrupts */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - srcr |= SSI_SRCR_RFEN0; - else - srcr |= SSI_SRCR_RFEN1; - sier |= SSI_SIER_RDMAE; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - SSI1_SRCR = srcr; - SSI1_SRCCR = srccr; - SSI1_SIER = sier; - } else { - SSI2_SRCR = srcr; - SSI2_SRCCR = srccr; - SSI2_SIER = sier; - } - - return 0; -} - -/* - * Should only be called when port is inactive (i.e. SSIEN = 0), - * although can be called multiple times by upper layers. - */ -int imx_ssi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - int ret; - - /* cant change any parameters when SSI is running */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - if (SSI1_SCR & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - } else { - if (SSI2_SCR & SSI_SCR_SSIEN) { - printk(KERN_WARNING "Warning ssi already enabled\n"); - return 0; - } - } - - /* - * Configure both tx and rx params with the same settings. This is - * really a harware restriction because SSI must be disabled until - * we can change those values. If there is an active audio stream in - * one direction, enabling the other direction with different - * settings would mean disturbing the running one. - */ - ret = imx_ssi_hw_tx_params(substream, params); - if (ret < 0) - return ret; - return imx_ssi_hw_rx_params(substream, params); -} - -int imx_ssi_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - pr_debug("%s\n", __func__); - - /* Enable clks here to follow SSI recommended init sequence */ - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { - ret = clk_enable(ssi_clk0); - if (ret < 0) - printk(KERN_ERR "Unable to enable ssi_clk0\n"); - } else { - ret = clk_enable(ssi_clk1); - if (ret < 0) - printk(KERN_ERR "Unable to enable ssi_clk1\n"); - } - - return 0; -} - -static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - u32 scr; - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - scr = SSI1_SCR; - else - scr = SSI2_SCR; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - scr |= SSI_SCR_TE | SSI_SCR_SSIEN; - else - scr |= SSI_SCR_RE | SSI_SCR_SSIEN; - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - scr &= ~SSI_SCR_TE; - else - scr &= ~SSI_SCR_RE; - break; - default: - return -EINVAL; - } - - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) - SSI1_SCR = scr; - else - SSI2_SCR = scr; - - return 0; -} - -static void imx_ssi_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - /* shutdown SSI if neither Tx or Rx is active */ - if (!cpu_dai->active) { - - if (cpu_dai->id == IMX_DAI_SSI0 || - cpu_dai->id == IMX_DAI_SSI2) { - - if (--ssi_active[SSI1_PORT] > 1) - return; - - SSI1_SCR = 0; - clk_disable(ssi_clk0); - } else { - if (--ssi_active[SSI2_PORT]) - return; - SSI2_SCR = 0; - clk_disable(ssi_clk1); - } - } -} - -#ifdef CONFIG_PM -static int imx_ssi_suspend(struct platform_device *dev, - struct snd_soc_dai *dai) -{ - return 0; -} - -static int imx_ssi_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - return 0; -} - -#else -#define imx_ssi_suspend NULL -#define imx_ssi_resume NULL -#endif - -#define IMX_SSI_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ - SNDRV_PCM_RATE_96000) - -#define IMX_SSI_BITS \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE) - -static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { - .startup = imx_ssi_startup, - .shutdown = imx_ssi_shutdown, - .trigger = imx_ssi_trigger, - .prepare = imx_ssi_prepare, - .hw_params = imx_ssi_hw_params, - .set_sysclk = imx_ssi_set_dai_sysclk, - .set_clkdiv = imx_ssi_set_dai_clkdiv, - .set_fmt = imx_ssi_set_dai_fmt, - .set_tdm_slot = imx_ssi_set_dai_tdm_slot, -}; - -struct snd_soc_dai imx_ssi_pcm_dai[] = { -{ - .name = "imx-i2s-1-0", - .id = IMX_DAI_SSI0, - .suspend = imx_ssi_suspend, - .resume = imx_ssi_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .ops = &imx_ssi_pcm_dai_ops, -}, -{ - .name = "imx-i2s-2-0", - .id = IMX_DAI_SSI1, - .playback = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .ops = &imx_ssi_pcm_dai_ops, -}, -{ - .name = "imx-i2s-1-1", - .id = IMX_DAI_SSI2, - .suspend = imx_ssi_suspend, - .resume = imx_ssi_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .ops = &imx_ssi_pcm_dai_ops, -}, -{ - .name = "imx-i2s-2-1", - .id = IMX_DAI_SSI3, - .playback = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .formats = IMX_SSI_BITS, - .rates = IMX_SSI_RATES,}, - .ops = &imx_ssi_pcm_dai_ops, -}, -}; -EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); - -static int __init imx_ssi_init(void) -{ - return snd_soc_register_dais(imx_ssi_pcm_dai, - ARRAY_SIZE(imx_ssi_pcm_dai)); -} - -static void __exit imx_ssi_exit(void) -{ - snd_soc_unregister_dais(imx_ssi_pcm_dai, - ARRAY_SIZE(imx_ssi_pcm_dai)); -} - -module_init(imx_ssi_init); -module_exit(imx_ssi_exit); -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com"); -MODULE_DESCRIPTION("i.MX ASoC I2S driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mxc-ssi.h b/sound/soc/imx/mxc-ssi.h deleted file mode 100644 index 12bbdc9..0000000 --- a/sound/soc/imx/mxc-ssi.h +++ /dev/null @@ -1,238 +0,0 @@ -/* - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _IMX_SSI_H -#define _IMX_SSI_H - -#include - -/* SSI regs definition - MOVE to /arch/arm/plat-mxc/include/mach/ when stable */ -#define SSI1_IO_BASE_ADDR IO_ADDRESS(SSI1_BASE_ADDR) -#define SSI2_IO_BASE_ADDR IO_ADDRESS(SSI2_BASE_ADDR) - -#define STX0 0x00 -#define STX1 0x04 -#define SRX0 0x08 -#define SRX1 0x0c -#define SCR 0x10 -#define SISR 0x14 -#define SIER 0x18 -#define STCR 0x1c -#define SRCR 0x20 -#define STCCR 0x24 -#define SRCCR 0x28 -#define SFCSR 0x2c -#define STR 0x30 -#define SOR 0x34 -#define SACNT 0x38 -#define SACADD 0x3c -#define SACDAT 0x40 -#define SATAG 0x44 -#define STMSK 0x48 -#define SRMSK 0x4c - -#define SSI1_STX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX0))) -#define SSI1_STX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX1))) -#define SSI1_SRX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX0))) -#define SSI1_SRX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX1))) -#define SSI1_SCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SCR))) -#define SSI1_SISR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SISR))) -#define SSI1_SIER (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SIER))) -#define SSI1_STCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCR))) -#define SSI1_SRCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCR))) -#define SSI1_STCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCCR))) -#define SSI1_SRCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCCR))) -#define SSI1_SFCSR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SFCSR))) -#define SSI1_STR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STR))) -#define SSI1_SOR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SOR))) -#define SSI1_SACNT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACNT))) -#define SSI1_SACADD (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACADD))) -#define SSI1_SACDAT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACDAT))) -#define SSI1_SATAG (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SATAG))) -#define SSI1_STMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STMSK))) -#define SSI1_SRMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRMSK))) - - -#define SSI2_STX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX0))) -#define SSI2_STX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX1))) -#define SSI2_SRX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX0))) -#define SSI2_SRX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX1))) -#define SSI2_SCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SCR))) -#define SSI2_SISR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SISR))) -#define SSI2_SIER (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SIER))) -#define SSI2_STCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCR))) -#define SSI2_SRCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCR))) -#define SSI2_STCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCCR))) -#define SSI2_SRCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCCR))) -#define SSI2_SFCSR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SFCSR))) -#define SSI2_STR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STR))) -#define SSI2_SOR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SOR))) -#define SSI2_SACNT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACNT))) -#define SSI2_SACADD (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACADD))) -#define SSI2_SACDAT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACDAT))) -#define SSI2_SATAG (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SATAG))) -#define SSI2_STMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STMSK))) -#define SSI2_SRMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRMSK))) - -#define SSI_SCR_CLK_IST (1 << 9) -#define SSI_SCR_TCH_EN (1 << 8) -#define SSI_SCR_SYS_CLK_EN (1 << 7) -#define SSI_SCR_I2S_MODE_NORM (0 << 5) -#define SSI_SCR_I2S_MODE_MSTR (1 << 5) -#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) -#define SSI_SCR_SYN (1 << 4) -#define SSI_SCR_NET (1 << 3) -#define SSI_SCR_RE (1 << 2) -#define SSI_SCR_TE (1 << 1) -#define SSI_SCR_SSIEN (1 << 0) - -#define SSI_SISR_CMDAU (1 << 18) -#define SSI_SISR_CMDDU (1 << 17) -#define SSI_SISR_RXT (1 << 16) -#define SSI_SISR_RDR1 (1 << 15) -#define SSI_SISR_RDR0 (1 << 14) -#define SSI_SISR_TDE1 (1 << 13) -#define SSI_SISR_TDE0 (1 << 12) -#define SSI_SISR_ROE1 (1 << 11) -#define SSI_SISR_ROE0 (1 << 10) -#define SSI_SISR_TUE1 (1 << 9) -#define SSI_SISR_TUE0 (1 << 8) -#define SSI_SISR_TFS (1 << 7) -#define SSI_SISR_RFS (1 << 6) -#define SSI_SISR_TLS (1 << 5) -#define SSI_SISR_RLS (1 << 4) -#define SSI_SISR_RFF1 (1 << 3) -#define SSI_SISR_RFF0 (1 << 2) -#define SSI_SISR_TFE1 (1 << 1) -#define SSI_SISR_TFE0 (1 << 0) - -#define SSI_SIER_RDMAE (1 << 22) -#define SSI_SIER_RIE (1 << 21) -#define SSI_SIER_TDMAE (1 << 20) -#define SSI_SIER_TIE (1 << 19) -#define SSI_SIER_CMDAU_EN (1 << 18) -#define SSI_SIER_CMDDU_EN (1 << 17) -#define SSI_SIER_RXT_EN (1 << 16) -#define SSI_SIER_RDR1_EN (1 << 15) -#define SSI_SIER_RDR0_EN (1 << 14) -#define SSI_SIER_TDE1_EN (1 << 13) -#define SSI_SIER_TDE0_EN (1 << 12) -#define SSI_SIER_ROE1_EN (1 << 11) -#define SSI_SIER_ROE0_EN (1 << 10) -#define SSI_SIER_TUE1_EN (1 << 9) -#define SSI_SIER_TUE0_EN (1 << 8) -#define SSI_SIER_TFS_EN (1 << 7) -#define SSI_SIER_RFS_EN (1 << 6) -#define SSI_SIER_TLS_EN (1 << 5) -#define SSI_SIER_RLS_EN (1 << 4) -#define SSI_SIER_RFF1_EN (1 << 3) -#define SSI_SIER_RFF0_EN (1 << 2) -#define SSI_SIER_TFE1_EN (1 << 1) -#define SSI_SIER_TFE0_EN (1 << 0) - -#define SSI_STCR_TXBIT0 (1 << 9) -#define SSI_STCR_TFEN1 (1 << 8) -#define SSI_STCR_TFEN0 (1 << 7) -#define SSI_STCR_TFDIR (1 << 6) -#define SSI_STCR_TXDIR (1 << 5) -#define SSI_STCR_TSHFD (1 << 4) -#define SSI_STCR_TSCKP (1 << 3) -#define SSI_STCR_TFSI (1 << 2) -#define SSI_STCR_TFSL (1 << 1) -#define SSI_STCR_TEFS (1 << 0) - -#define SSI_SRCR_RXBIT0 (1 << 9) -#define SSI_SRCR_RFEN1 (1 << 8) -#define SSI_SRCR_RFEN0 (1 << 7) -#define SSI_SRCR_RFDIR (1 << 6) -#define SSI_SRCR_RXDIR (1 << 5) -#define SSI_SRCR_RSHFD (1 << 4) -#define SSI_SRCR_RSCKP (1 << 3) -#define SSI_SRCR_RFSI (1 << 2) -#define SSI_SRCR_RFSL (1 << 1) -#define SSI_SRCR_REFS (1 << 0) - -#define SSI_STCCR_DIV2 (1 << 18) -#define SSI_STCCR_PSR (1 << 15) -#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) -#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) -#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) -#define SSI_STCCR_WL_MASK (0xf << 13) -#define SSI_STCCR_DC_MASK (0x1f << 8) -#define SSI_STCCR_PM_MASK (0xff << 0) - -#define SSI_SRCCR_DIV2 (1 << 18) -#define SSI_SRCCR_PSR (1 << 15) -#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) -#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) -#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) -#define SSI_SRCCR_WL_MASK (0xf << 13) -#define SSI_SRCCR_DC_MASK (0x1f << 8) -#define SSI_SRCCR_PM_MASK (0xff << 0) - - -#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) -#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) -#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) -#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) -#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) -#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) -#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) -#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) - -#define SSI_STR_TEST (1 << 15) -#define SSI_STR_RCK2TCK (1 << 14) -#define SSI_STR_RFS2TFS (1 << 13) -#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) -#define SSI_STR_TXD2RXD (1 << 7) -#define SSI_STR_TCK2RCK (1 << 6) -#define SSI_STR_TFS2RFS (1 << 5) -#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) - -#define SSI_SOR_CLKOFF (1 << 6) -#define SSI_SOR_RX_CLR (1 << 5) -#define SSI_SOR_TX_CLR (1 << 4) -#define SSI_SOR_INIT (1 << 3) -#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) -#define SSI_SOR_SYNRST (1 << 0) - -#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) -#define SSI_SACNT_WR (x << 4) -#define SSI_SACNT_RD (x << 3) -#define SSI_SACNT_TIF (x << 2) -#define SSI_SACNT_FV (x << 1) -#define SSI_SACNT_AC97EN (x << 0) - -/* Watermarks for FIFO's */ -#define TXFIFO_WATERMARK 0x4 -#define RXFIFO_WATERMARK 0x4 - -/* i.MX DAI SSP ID's */ -#define IMX_DAI_SSI0 0 /* SSI1 FIFO 0 */ -#define IMX_DAI_SSI1 1 /* SSI1 FIFO 1 */ -#define IMX_DAI_SSI2 2 /* SSI2 FIFO 0 */ -#define IMX_DAI_SSI3 3 /* SSI2 FIFO 1 */ - -/* SSI clock sources */ -#define IMX_SSP_SYS_CLK 0 - -/* SSI audio dividers */ -#define IMX_SSI_TX_DIV_2 0 -#define IMX_SSI_TX_DIV_PSR 1 -#define IMX_SSI_TX_DIV_PM 2 -#define IMX_SSI_RX_DIV_2 3 -#define IMX_SSI_RX_DIV_PSR 4 -#define IMX_SSI_RX_DIV_PM 5 - - -/* SSI Div 2 */ -#define IMX_SSI_DIV_2_OFF (~SSI_STCCR_DIV2) -#define IMX_SSI_DIV_2_ON SSI_STCCR_DIV2 - -extern struct snd_soc_dai imx_ssi_pcm_dai[4]; -extern int get_ssi_clk(int ssi, struct device *dev); -extern void put_ssi_clk(int ssi); -#endif -- cgit v0.10.2 From b05f5c13d5bc2fa9945c9534f8881396555290a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 17 Jan 2010 16:45:06 +0000 Subject: ASoC: Mark new i.MX drivers as BROKEN until arch/arm merged Currently they don't build due to cross tree dependencies, they will be reenabled once the arch/arm side has merged. Signed-off-by: Mark Brown diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 84a25e6..5f006f0d 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,6 +1,6 @@ config SND_IMX_SOC tristate "SoC Audio for Freecale i.MX CPUs" - depends on ARCH_MXC + depends on ARCH_MXC && BROKEN select SND_PCM select FIQ select SND_SOC_AC97_BUS -- cgit v0.10.2 From 3e879d7bac705be4813a0ec9560cbe31db4b269f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:49:50 +0100 Subject: ALSA: pcm - Remove unneeded ifdef pgprot_noncached Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a870fe69..5df0d21 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3162,9 +3162,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, long size; unsigned long offset; -#ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); -#endif area->vm_flags |= VM_IO; size = area->vm_end - area->vm_start; offset = area->vm_pgoff << PAGE_SHIFT; -- cgit v0.10.2 From c32d977b8157bf67cdf47729ce7dd054a26eb534 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:58:57 +0100 Subject: ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffers pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd need non-cached behavior more or less, even for the intermediate ring- buffers. Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co. Signed-off-by: Takashi Iwai diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1d4ca2a..aabf48b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1021,6 +1021,10 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif +int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, + struct vm_area_struct *area); +#define snd_pcm_lib_mmap_vmalloc snd_pcm_lib_mmap_noncached + static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) { *max = dma < 4 ? 64 * 1024 : 128 * 1024; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 5df0d21..88fff44 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3176,6 +3176,15 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); #endif /* SNDRV_PCM_INFO_MMAP */ +/* mmap callback with pgprot_noncached */ +int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); + return snd_pcm_default_mmap(substream, area); +} +EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached); + /* * mmap DMA buffer */ diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index c8385d2..35a2f71 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -905,6 +905,7 @@ static struct snd_pcm_ops vx_pcm_playback_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; @@ -1125,6 +1126,7 @@ static struct snd_pcm_ops vx_pcm_capture_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9b486be..6aff217 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -691,6 +691,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -703,6 +704,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -715,6 +717,7 @@ static struct snd_pcm_ops snd_sgio2audio_capture_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; /* diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 0afa683..0d668f4 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -277,6 +277,7 @@ static struct snd_pcm_ops pdacf_pcm_capture_ops = { .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index 16dc7bd..4f4ccdf 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -911,6 +911,7 @@ static struct snd_pcm_ops capture_pcm_ops = { .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops playback_pcm_ops = { @@ -923,6 +924,7 @@ static struct snd_pcm_ops playback_pcm_ops = { .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct uac_format_type_i_discrete_descriptor * diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4ada98e..b8e0b8f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1997,6 +1997,7 @@ static struct snd_pcm_ops snd_usb_playback_ops = { .trigger = snd_usb_pcm_playback_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_usb_capture_ops = { @@ -2009,6 +2010,7 @@ static struct snd_pcm_ops snd_usb_capture_ops = { .trigger = snd_usb_pcm_capture_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; -- cgit v0.10.2 From a32f66746c635ebf2341d99b3d4c0cc1c11b2cbf Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 18 Jan 2010 15:40:56 +0100 Subject: sound: seq_timer: simplify snd_seq_timer_set_tick_resolution() parameters As snd_seq_timer_set_tick_resolution() is always called with the same three fields of struct snd_seq_timer, it suffices to give that as the only parameter. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index f745c31..160b1bd 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -33,22 +33,21 @@ #define SKEW_BASE 0x10000 /* 16bit shift */ -static void snd_seq_timer_set_tick_resolution(struct snd_seq_timer_tick *tick, - int tempo, int ppq) +static void snd_seq_timer_set_tick_resolution(struct snd_seq_timer *tmr) { - if (tempo < 1000000) - tick->resolution = (tempo * 1000) / ppq; + if (tmr->tempo < 1000000) + tmr->tick.resolution = (tmr->tempo * 1000) / tmr->ppq; else { /* might overflow.. */ unsigned int s; - s = tempo % ppq; - s = (s * 1000) / ppq; - tick->resolution = (tempo / ppq) * 1000; - tick->resolution += s; + s = tmr->tempo % tmr->ppq; + s = (s * 1000) / tmr->ppq; + tmr->tick.resolution = (tmr->tempo / tmr->ppq) * 1000; + tmr->tick.resolution += s; } - if (tick->resolution <= 0) - tick->resolution = 1; - snd_seq_timer_update_tick(tick, 0); + if (tmr->tick.resolution <= 0) + tmr->tick.resolution = 1; + snd_seq_timer_update_tick(&tmr->tick, 0); } /* create new timer (constructor) */ @@ -96,7 +95,7 @@ void snd_seq_timer_defaults(struct snd_seq_timer * tmr) /* setup defaults */ tmr->ppq = 96; /* 96 PPQ */ tmr->tempo = 500000; /* 120 BPM */ - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); tmr->running = 0; tmr->type = SNDRV_SEQ_TIMER_ALSA; @@ -180,7 +179,7 @@ int snd_seq_timer_set_tempo(struct snd_seq_timer * tmr, int tempo) spin_lock_irqsave(&tmr->lock, flags); if ((unsigned int)tempo != tmr->tempo) { tmr->tempo = tempo; - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); } spin_unlock_irqrestore(&tmr->lock, flags); return 0; @@ -205,7 +204,7 @@ int snd_seq_timer_set_ppq(struct snd_seq_timer * tmr, int ppq) } tmr->ppq = ppq; - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); spin_unlock_irqrestore(&tmr->lock, flags); return 0; } -- cgit v0.10.2 From d1db38c015a392b0ea8c15ab95abb3ee768b8d47 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 18 Jan 2010 15:44:04 +0100 Subject: sound: virtuoso: add Xonar DS support Add experimental support for the Asus Xonar DS. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8923597..3579e82 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1923,7 +1923,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ------------------- Module for sound cards based on the Asus AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), Essence ST + i.e., Xonar D1, DX, D2, D2X, DS, HDAV1.3 (Deluxe), Essence ST (Deluxe) and Essence STX. This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 351654c..1298c68 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -789,6 +789,7 @@ config SND_VIRTUOSO Say Y here to include support for sound cards based on the Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, Essence ST (Deluxe), and Essence STX. + Support for the DS is experimental. Support for the HDAV1.3 (Deluxe) is very experimental. To compile this driver as a module, choose M here: the module diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index 389941c..acd8f15 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -2,7 +2,7 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o snd-hifier-objs := hifier.o snd-oxygen-objs := oxygen.o snd-virtuoso-objs := virtuoso.o xonar_lib.o \ - xonar_pcm179x.o xonar_cs43xx.o xonar_hdmi.o + xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o obj-$(CONFIG_SND_HIFIER) += snd-hifier.o diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 6accaf9..563b6f5 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -49,6 +49,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x834f) }, { OXYGEN_PCI_SUBID(0x1043, 0x835c) }, { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, + { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; @@ -61,6 +62,8 @@ static int __devinit get_xonar_model(struct oxygen *chip, return 0; if (get_xonar_cs43xx_model(chip, id) >= 0) return 0; + if (get_xonar_wm87x6_model(chip, id) >= 0) + return 0; return -EINVAL; } diff --git a/sound/pci/oxygen/wm8766.h b/sound/pci/oxygen/wm8766.h new file mode 100644 index 0000000..e0e849a --- /dev/null +++ b/sound/pci/oxygen/wm8766.h @@ -0,0 +1,73 @@ +#ifndef WM8766_H_INCLUDED +#define WM8766_H_INCLUDED + +#define WM8766_LDA1 0x00 +#define WM8766_RDA1 0x01 +#define WM8766_DAC_CTRL 0x02 +#define WM8766_INT_CTRL 0x03 +#define WM8766_LDA2 0x04 +#define WM8766_RDA2 0x05 +#define WM8766_LDA3 0x06 +#define WM8766_RDA3 0x07 +#define WM8766_MASTDA 0x08 +#define WM8766_DAC_CTRL2 0x09 +#define WM8766_DAC_CTRL3 0x0a +#define WM8766_MUTE1 0x0c +#define WM8766_MUTE2 0x0f +#define WM8766_RESET 0x1f + +/* LDAx/RDAx/MASTDA */ +#define WM8766_ATT_MASK 0x0ff +#define WM8766_UPDATE 0x100 +/* DAC_CTRL */ +#define WM8766_MUTEALL 0x001 +#define WM8766_DEEMPALL 0x002 +#define WM8766_PWDN 0x004 +#define WM8766_ATC 0x008 +#define WM8766_IZD 0x010 +#define WM8766_PL_LEFT_MASK 0x060 +#define WM8766_PL_LEFT_MUTE 0x000 +#define WM8766_PL_LEFT_LEFT 0x020 +#define WM8766_PL_LEFT_RIGHT 0x040 +#define WM8766_PL_LEFT_LRMIX 0x060 +#define WM8766_PL_RIGHT_MASK 0x180 +#define WM8766_PL_RIGHT_MUTE 0x000 +#define WM8766_PL_RIGHT_LEFT 0x080 +#define WM8766_PL_RIGHT_RIGHT 0x100 +#define WM8766_PL_RIGHT_LRMIX 0x180 +/* INT_CTRL */ +#define WM8766_FMT_MASK 0x003 +#define WM8766_FMT_RJUST 0x000 +#define WM8766_FMT_LJUST 0x001 +#define WM8766_FMT_I2S 0x002 +#define WM8766_FMT_DSP 0x003 +#define WM8766_LRP 0x004 +#define WM8766_BCP 0x008 +#define WM8766_IWL_MASK 0x030 +#define WM8766_IWL_16 0x000 +#define WM8766_IWL_20 0x010 +#define WM8766_IWL_24 0x020 +#define WM8766_IWL_32 0x030 +#define WM8766_PHASE_MASK 0x1c0 +/* DAC_CTRL2 */ +#define WM8766_ZCD 0x001 +#define WM8766_DZFM_MASK 0x006 +#define WM8766_DMUTE_MASK 0x038 +#define WM8766_DEEMP_MASK 0x1c0 +/* DAC_CTRL3 */ +#define WM8766_DACPD_MASK 0x00e +#define WM8766_PWRDNALL 0x010 +#define WM8766_MS 0x020 +#define WM8766_RATE_MASK 0x1c0 +#define WM8766_RATE_128 0x000 +#define WM8766_RATE_192 0x040 +#define WM8766_RATE_256 0x080 +#define WM8766_RATE_384 0x0c0 +#define WM8766_RATE_512 0x100 +#define WM8766_RATE_768 0x140 +/* MUTE1 */ +#define WM8766_MPD1 0x040 +/* MUTE2 */ +#define WM8766_MPD2 0x020 + +#endif diff --git a/sound/pci/oxygen/wm8776.h b/sound/pci/oxygen/wm8776.h new file mode 100644 index 0000000..1a96f56 --- /dev/null +++ b/sound/pci/oxygen/wm8776.h @@ -0,0 +1,177 @@ +#ifndef WM8776_H_INCLUDED +#define WM8776_H_INCLUDED + +/* + * the following register names are from: + * wm8776.h -- WM8776 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#define WM8776_HPLVOL 0x00 +#define WM8776_HPRVOL 0x01 +#define WM8776_HPMASTER 0x02 +#define WM8776_DACLVOL 0x03 +#define WM8776_DACRVOL 0x04 +#define WM8776_DACMASTER 0x05 +#define WM8776_PHASESWAP 0x06 +#define WM8776_DACCTRL1 0x07 +#define WM8776_DACMUTE 0x08 +#define WM8776_DACCTRL2 0x09 +#define WM8776_DACIFCTRL 0x0a +#define WM8776_ADCIFCTRL 0x0b +#define WM8776_MSTRCTRL 0x0c +#define WM8776_PWRDOWN 0x0d +#define WM8776_ADCLVOL 0x0e +#define WM8776_ADCRVOL 0x0f +#define WM8776_ALCCTRL1 0x10 +#define WM8776_ALCCTRL2 0x11 +#define WM8776_ALCCTRL3 0x12 +#define WM8776_NOISEGATE 0x13 +#define WM8776_LIMITER 0x14 +#define WM8776_ADCMUX 0x15 +#define WM8776_OUTMUX 0x16 +#define WM8776_RESET 0x17 + + +/* HPLVOL/HPRVOL/HPMASTER */ +#define WM8776_HPATT_MASK 0x07f +#define WM8776_HPZCEN 0x080 +#define WM8776_UPDATE 0x100 + +/* DACLVOL/DACRVOL/DACMASTER */ +#define WM8776_DATT_MASK 0x0ff +/*#define WM8776_UPDATE 0x100*/ + +/* PHASESWAP */ +#define WM8776_PH_MASK 0x003 + +/* DACCTRL1 */ +#define WM8776_DZCEN 0x001 +#define WM8776_ATC 0x002 +#define WM8776_IZD 0x004 +#define WM8776_TOD 0x008 +#define WM8776_PL_LEFT_MASK 0x030 +#define WM8776_PL_LEFT_MUTE 0x000 +#define WM8776_PL_LEFT_LEFT 0x010 +#define WM8776_PL_LEFT_RIGHT 0x020 +#define WM8776_PL_LEFT_LRMIX 0x030 +#define WM8776_PL_RIGHT_MASK 0x0c0 +#define WM8776_PL_RIGHT_MUTE 0x000 +#define WM8776_PL_RIGHT_LEFT 0x040 +#define WM8776_PL_RIGHT_RIGHT 0x080 +#define WM8776_PL_RIGHT_LRMIX 0x0c0 + +/* DACMUTE */ +#define WM8776_DMUTE 0x001 + +/* DACCTRL2 */ +#define WM8776_DEEMPH 0x001 +#define WM8776_DZFM_MASK 0x006 +#define WM8776_DZFM_NONE 0x000 +#define WM8776_DZFM_LR 0x002 +#define WM8776_DZFM_BOTH 0x004 +#define WM8776_DZFM_EITHER 0x006 + +/* DACIFCTRL */ +#define WM8776_DACFMT_MASK 0x003 +#define WM8776_DACFMT_RJUST 0x000 +#define WM8776_DACFMT_LJUST 0x001 +#define WM8776_DACFMT_I2S 0x002 +#define WM8776_DACFMT_DSP 0x003 +#define WM8776_DACLRP 0x004 +#define WM8776_DACBCP 0x008 +#define WM8776_DACWL_MASK 0x030 +#define WM8776_DACWL_16 0x000 +#define WM8776_DACWL_20 0x010 +#define WM8776_DACWL_24 0x020 +#define WM8776_DACWL_32 0x030 + +/* ADCIFCTRL */ +#define WM8776_ADCFMT_MASK 0x003 +#define WM8776_ADCFMT_RJUST 0x000 +#define WM8776_ADCFMT_LJUST 0x001 +#define WM8776_ADCFMT_I2S 0x002 +#define WM8776_ADCFMT_DSP 0x003 +#define WM8776_ADCLRP 0x004 +#define WM8776_ADCBCP 0x008 +#define WM8776_ADCWL_MASK 0x030 +#define WM8776_ADCWL_16 0x000 +#define WM8776_ADCWL_20 0x010 +#define WM8776_ADCWL_24 0x020 +#define WM8776_ADCWL_32 0x030 +#define WM8776_ADCMCLK 0x040 +#define WM8776_ADCHPD 0x100 + +/* MSTRCTRL */ +#define WM8776_ADCRATE_MASK 0x007 +#define WM8776_ADCRATE_256 0x002 +#define WM8776_ADCRATE_384 0x003 +#define WM8776_ADCRATE_512 0x004 +#define WM8776_ADCRATE_768 0x005 +#define WM8776_ADCOSR 0x008 +#define WM8776_DACRATE_MASK 0x070 +#define WM8776_DACRATE_128 0x000 +#define WM8776_DACRATE_192 0x010 +#define WM8776_DACRATE_256 0x020 +#define WM8776_DACRATE_384 0x030 +#define WM8776_DACRATE_512 0x040 +#define WM8776_DACRATE_768 0x050 +#define WM8776_DACMS 0x080 +#define WM8776_ADCMS 0x100 + +/* PWRDOWN */ +#define WM8776_PDWN 0x001 +#define WM8776_ADCPD 0x002 +#define WM8776_DACPD 0x004 +#define WM8776_HPPD 0x008 +#define WM8776_AINPD 0x040 + +/* ADCLVOL/ADCRVOL */ +#define WM8776_AGMASK 0x0ff +#define WM8776_ZCA 0x100 + +/* ALCCTRL1 */ +#define WM8776_LCT_MASK 0x00f +#define WM8776_MAXGAIN_MASK 0x070 +#define WM8776_LCSEL_MASK 0x180 +#define WM8776_LCSEL_LIMITER 0x000 +#define WM8776_LCSEL_ALC_RIGHT 0x080 +#define WM8776_LCSEL_ALC_LEFT 0x100 +#define WM8776_LCSEL_ALC_STEREO 0x180 + +/* ALCCTRL2 */ +#define WM8776_HLD_MASK 0x00f +#define WM8776_ALCZC 0x080 +#define WM8776_LCEN 0x100 + +/* ALCCTRL3 */ +#define WM8776_ATK_MASK 0x00f +#define WM8776_DCY_MASK 0x0f0 + +/* NOISEGATE */ +#define WM8776_NGAT 0x001 +#define WM8776_NGTH_MASK 0x01c + +/* LIMITER */ +#define WM8776_MAXATTEN_MASK 0x00f +#define WM8776_TRANWIN_MASK 0x070 + +/* ADCMUX */ +#define WM8776_AMX_MASK 0x01f +#define WM8776_MUTERA 0x040 +#define WM8776_MUTELA 0x080 +#define WM8776_LRBOTH 0x100 + +/* OUTMUX */ +#define WM8776_MX_DAC 0x001 +#define WM8776_MX_AUX 0x002 +#define WM8776_MX_BYPASS 0x004 + +#endif diff --git a/sound/pci/oxygen/xonar.h b/sound/pci/oxygen/xonar.h index 89b3ed8..b35343b 100644 --- a/sound/pci/oxygen/xonar.h +++ b/sound/pci/oxygen/xonar.h @@ -35,6 +35,8 @@ int get_xonar_pcm179x_model(struct oxygen *chip, const struct pci_device_id *id); int get_xonar_cs43xx_model(struct oxygen *chip, const struct pci_device_id *id); +int get_xonar_wm87x6_model(struct oxygen *chip, + const struct pci_device_id *id); /* HDMI helper functions */ diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c new file mode 100644 index 0000000..7754db1 --- /dev/null +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -0,0 +1,1021 @@ +/* + * card driver for models with WM8776/WM8766 DACs (Xonar DS) + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see . + */ + +/* + * Xonar DS + * -------- + * + * CMI8788: + * + * SPI 0 -> WM8766 (surround, center/LFE, back) + * SPI 1 -> WM8776 (front, input) + * + * GPIO 4 <- headphone detect + * GPIO 6 -> route input jack to input 1/2 (1/0) + * GPIO 7 -> enable output to speakers + * GPIO 8 -> enable output to speakers + */ + +#include +#include +#include +#include +#include +#include +#include +#include "xonar.h" +#include "wm8776.h" +#include "wm8766.h" + +#define GPIO_DS_HP_DETECT 0x0010 +#define GPIO_DS_INPUT_ROUTE 0x0040 +#define GPIO_DS_OUTPUT_ENABLE 0x0180 + +#define LC_CONTROL_LIMITER 0x40000000 +#define LC_CONTROL_ALC 0x20000000 + +struct xonar_wm87x6 { + struct xonar_generic generic; + u16 wm8776_regs[0x17]; + u16 wm8766_regs[0x10]; + struct snd_kcontrol *lc_controls[13]; +}; + +static void wm8776_write(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | + (1 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_LO, + (reg << 9) | value); + if (reg < ARRAY_SIZE(data->wm8776_regs)) { + if (reg >= WM8776_HPLVOL || reg <= WM8776_DACMASTER) + value &= ~WM8776_UPDATE; + data->wm8776_regs[reg] = value; + } +} + +static void wm8776_write_cached(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + if (reg >= ARRAY_SIZE(data->wm8776_regs) || + value != data->wm8776_regs[reg]) + wm8776_write(chip, reg, value); +} + +static void wm8766_write(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | + (0 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_LO, + (reg << 9) | value); + if (reg < ARRAY_SIZE(data->wm8766_regs)) + data->wm8766_regs[reg] = value; +} + +static void wm8766_write_cached(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + if (reg >= ARRAY_SIZE(data->wm8766_regs) || + value != data->wm8766_regs[reg]) { + if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) || + (reg >= WM8766_LDA2 && reg <= WM8766_MASTDA)) + value &= ~WM8766_UPDATE; + wm8766_write(chip, reg, value); + } +} + +static void wm8776_registers_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + wm8776_write(chip, WM8776_RESET, 0); + wm8776_write(chip, WM8776_DACCTRL1, WM8776_DZCEN | + WM8776_PL_LEFT_LEFT | WM8776_PL_RIGHT_RIGHT); + wm8776_write(chip, WM8776_DACMUTE, chip->dac_mute ? WM8776_DMUTE : 0); + wm8776_write(chip, WM8776_DACIFCTRL, + WM8776_DACFMT_LJUST | WM8776_DACWL_24); + wm8776_write(chip, WM8776_ADCIFCTRL, + data->wm8776_regs[WM8776_ADCIFCTRL]); + wm8776_write(chip, WM8776_MSTRCTRL, data->wm8776_regs[WM8776_MSTRCTRL]); + wm8776_write(chip, WM8776_PWRDOWN, data->wm8776_regs[WM8776_PWRDOWN]); + wm8776_write(chip, WM8776_HPLVOL, data->wm8776_regs[WM8776_HPLVOL]); + wm8776_write(chip, WM8776_HPRVOL, data->wm8776_regs[WM8776_HPRVOL] | + WM8776_UPDATE); + wm8776_write(chip, WM8776_ADCLVOL, data->wm8776_regs[WM8776_ADCLVOL]); + wm8776_write(chip, WM8776_ADCRVOL, data->wm8776_regs[WM8776_ADCRVOL]); + wm8776_write(chip, WM8776_ADCMUX, data->wm8776_regs[WM8776_ADCMUX]); + wm8776_write(chip, WM8776_DACLVOL, chip->dac_volume[0]); + wm8776_write(chip, WM8776_DACRVOL, chip->dac_volume[1] | WM8776_UPDATE); +} + +static void wm8766_registers_init(struct oxygen *chip) +{ + wm8766_write(chip, WM8766_RESET, 0); + wm8766_write(chip, WM8766_INT_CTRL, WM8766_FMT_LJUST | WM8766_IWL_24); + wm8766_write(chip, WM8766_DAC_CTRL2, + WM8766_ZCD | (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); + wm8766_write(chip, WM8766_LDA1, chip->dac_volume[2]); + wm8766_write(chip, WM8766_RDA1, chip->dac_volume[3]); + wm8766_write(chip, WM8766_LDA2, chip->dac_volume[4]); + wm8766_write(chip, WM8766_RDA2, chip->dac_volume[5]); + wm8766_write(chip, WM8766_LDA3, chip->dac_volume[6]); + wm8766_write(chip, WM8766_RDA3, chip->dac_volume[7] | WM8766_UPDATE); +} + +static void wm8776_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->wm8776_regs[WM8776_HPLVOL] = (0x79 - 60) | WM8776_HPZCEN; + data->wm8776_regs[WM8776_HPRVOL] = (0x79 - 60) | WM8776_HPZCEN; + data->wm8776_regs[WM8776_ADCIFCTRL] = + WM8776_ADCFMT_LJUST | WM8776_ADCWL_24 | WM8776_ADCMCLK; + data->wm8776_regs[WM8776_MSTRCTRL] = + WM8776_ADCRATE_256 | WM8776_DACRATE_256; + data->wm8776_regs[WM8776_PWRDOWN] = WM8776_HPPD; + data->wm8776_regs[WM8776_ADCLVOL] = 0xa5 | WM8776_ZCA; + data->wm8776_regs[WM8776_ADCRVOL] = 0xa5 | WM8776_ZCA; + data->wm8776_regs[WM8776_ADCMUX] = 0x001; + wm8776_registers_init(chip); +} + +static void xonar_ds_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->generic.anti_pop_delay = 300; + data->generic.output_enable_bit = GPIO_DS_OUTPUT_ENABLE; + + wm8776_init(chip); + wm8766_registers_init(chip); + + oxygen_write16_masked(chip, OXYGEN_GPIO_CONTROL, GPIO_DS_INPUT_ROUTE, + GPIO_DS_HP_DETECT | GPIO_DS_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DS_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, GPIO_DS_HP_DETECT); + chip->interrupt_mask |= OXYGEN_INT_GPIO; + + xonar_enable_output(chip); + + snd_component_add(chip->card, "WM8776"); + snd_component_add(chip->card, "WM8766"); +} + +static void xonar_ds_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + +static void xonar_ds_suspend(struct oxygen *chip) +{ + xonar_ds_cleanup(chip); +} + +static void xonar_ds_resume(struct oxygen *chip) +{ + wm8776_registers_init(chip); + wm8766_registers_init(chip); + xonar_enable_output(chip); +} + +static void wm8776_adc_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware) +{ + if (channel == PCM_A) { + hardware->rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + hardware->rate_max = 96000; + } +} + +static void set_wm87x6_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ +} + +static void set_wm8776_adc_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + u16 reg; + + reg = WM8776_ADCRATE_256 | WM8776_DACRATE_256; + if (params_rate(params) > 48000) + reg |= WM8776_ADCOSR; + wm8776_write_cached(chip, WM8776_MSTRCTRL, reg); +} + +static void update_wm8776_volume(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + u8 to_change; + + if (chip->dac_volume[0] == chip->dac_volume[1]) { + if (chip->dac_volume[0] != data->wm8776_regs[WM8776_DACLVOL] || + chip->dac_volume[1] != data->wm8776_regs[WM8776_DACRVOL]) { + wm8776_write(chip, WM8776_DACMASTER, + chip->dac_volume[0] | WM8776_UPDATE); + data->wm8776_regs[WM8776_DACLVOL] = chip->dac_volume[0]; + data->wm8776_regs[WM8776_DACRVOL] = chip->dac_volume[0]; + } + } else { + to_change = (chip->dac_volume[0] != + data->wm8776_regs[WM8776_DACLVOL]) << 0; + to_change |= (chip->dac_volume[1] != + data->wm8776_regs[WM8776_DACLVOL]) << 1; + if (to_change & 1) + wm8776_write(chip, WM8776_DACLVOL, chip->dac_volume[0] | + ((to_change & 2) ? 0 : WM8776_UPDATE)); + if (to_change & 2) + wm8776_write(chip, WM8776_DACRVOL, + chip->dac_volume[1] | WM8776_UPDATE); + } +} + +static void update_wm87x6_volume(struct oxygen *chip) +{ + static const u8 wm8766_regs[6] = { + WM8766_LDA1, WM8766_RDA1, + WM8766_LDA2, WM8766_RDA2, + WM8766_LDA3, WM8766_RDA3, + }; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + u8 to_change; + + update_wm8776_volume(chip); + if (chip->dac_volume[2] == chip->dac_volume[3] && + chip->dac_volume[2] == chip->dac_volume[4] && + chip->dac_volume[2] == chip->dac_volume[5] && + chip->dac_volume[2] == chip->dac_volume[6] && + chip->dac_volume[2] == chip->dac_volume[7]) { + to_change = 0; + for (i = 0; i < 6; ++i) + if (chip->dac_volume[2] != + data->wm8766_regs[wm8766_regs[i]]) + to_change = 1; + if (to_change) { + wm8766_write(chip, WM8766_MASTDA, + chip->dac_volume[2] | WM8766_UPDATE); + for (i = 0; i < 6; ++i) + data->wm8766_regs[wm8766_regs[i]] = + chip->dac_volume[2]; + } + } else { + to_change = 0; + for (i = 0; i < 6; ++i) + to_change |= (chip->dac_volume[2 + i] != + data->wm8766_regs[wm8766_regs[i]]) << i; + for (i = 0; i < 6; ++i) + if (to_change & (1 << i)) + wm8766_write(chip, wm8766_regs[i], + chip->dac_volume[2 + i] | + ((to_change & (0x3e << i)) + ? 0 : WM8766_UPDATE)); + } +} + +static void update_wm8776_mute(struct oxygen *chip) +{ + wm8776_write_cached(chip, WM8776_DACMUTE, + chip->dac_mute ? WM8776_DMUTE : 0); +} + +static void update_wm87x6_mute(struct oxygen *chip) +{ + update_wm8776_mute(chip); + wm8766_write_cached(chip, WM8766_DAC_CTRL2, WM8766_ZCD | + (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); +} + +static void xonar_ds_gpio_changed(struct oxygen *chip) +{ + u16 bits; + + bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); + snd_printk(KERN_INFO "HP detect: %d\n", !!(bits & GPIO_DS_HP_DETECT)); +} + +static int wm8776_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u16 bit = ctl->private_value & 0xffff; + unsigned int reg_index = (ctl->private_value >> 16) & 0xff; + bool invert = (ctl->private_value >> 24) & 1; + + value->value.integer.value[0] = + ((data->wm8776_regs[reg_index] & bit) != 0) ^ invert; + return 0; +} + +static int wm8776_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u16 bit = ctl->private_value & 0xffff; + u16 reg_value; + unsigned int reg_index = (ctl->private_value >> 16) & 0xff; + bool invert = (ctl->private_value >> 24) & 1; + int changed; + + mutex_lock(&chip->mutex); + reg_value = data->wm8776_regs[reg_index] & ~bit; + if (value->value.integer.value[0] ^ invert) + reg_value |= bit; + changed = reg_value != data->wm8776_regs[reg_index]; + if (changed) + wm8776_write(chip, reg_index, reg_value); + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_field_enum_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const hld[16] = { + "0 ms", "2.67 ms", "5.33 ms", "10.6 ms", + "21.3 ms", "42.7 ms", "85.3 ms", "171 ms", + "341 ms", "683 ms", "1.37 s", "2.73 s", + "5.46 s", "10.9 s", "21.8 s", "43.7 s", + }; + static const char *const atk_lim[11] = { + "0.25 ms", "0.5 ms", "1 ms", "2 ms", + "4 ms", "8 ms", "16 ms", "32 ms", + "64 ms", "128 ms", "256 ms", + }; + static const char *const atk_alc[11] = { + "8.40 ms", "16.8 ms", "33.6 ms", "67.2 ms", + "134 ms", "269 ms", "538 ms", "1.08 s", + "2.15 s", "4.3 s", "8.6 s", + }; + static const char *const dcy_lim[11] = { + "1.2 ms", "2.4 ms", "4.8 ms", "9.6 ms", + "19.2 ms", "38.4 ms", "76.8 ms", "154 ms", + "307 ms", "614 ms", "1.23 s", + }; + static const char *const dcy_alc[11] = { + "33.5 ms", "67.0 ms", "134 ms", "268 ms", + "536 ms", "1.07 s", "2.14 s", "4.29 s", + "8.58 s", "17.2 s", "34.3 s", + }; + static const char *const tranwin[8] = { + "0 us", "62.5 us", "125 us", "250 us", + "500 us", "1 ms", "2 ms", "4 ms", + }; + u8 max; + const char *const *names; + + max = (ctl->private_value >> 12) & 0xf; + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = max + 1; + if (info->value.enumerated.item > max) + info->value.enumerated.item = max; + switch ((ctl->private_value >> 24) & 0x1f) { + case WM8776_ALCCTRL2: + names = hld; + break; + case WM8776_ALCCTRL3: + if (((ctl->private_value >> 20) & 0xf) == 0) { + if (ctl->private_value & LC_CONTROL_LIMITER) + names = atk_lim; + else + names = atk_alc; + } else { + if (ctl->private_value & LC_CONTROL_LIMITER) + names = dcy_lim; + else + names = dcy_alc; + } + break; + case WM8776_LIMITER: + names = tranwin; + break; + default: + return -ENXIO; + } + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int wm8776_field_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = (ctl->private_value >> 8) & 0xf; + info->value.integer.max = (ctl->private_value >> 12) & 0xf; + return 0; +} + +static void wm8776_field_set_from_ctl(struct snd_kcontrol *ctl) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int value, reg_index, mode; + u8 min, max, shift; + u16 mask, reg_value; + bool invert; + + if ((data->wm8776_regs[WM8776_ALCCTRL1] & WM8776_LCSEL_MASK) == + WM8776_LCSEL_LIMITER) + mode = LC_CONTROL_LIMITER; + else + mode = LC_CONTROL_ALC; + if (!(ctl->private_value & mode)) + return; + + value = ctl->private_value & 0xf; + min = (ctl->private_value >> 8) & 0xf; + max = (ctl->private_value >> 12) & 0xf; + mask = (ctl->private_value >> 16) & 0xf; + shift = (ctl->private_value >> 20) & 0xf; + reg_index = (ctl->private_value >> 24) & 0x1f; + invert = (ctl->private_value >> 29) & 0x1; + + if (invert) + value = max - (value - min); + reg_value = data->wm8776_regs[reg_index]; + reg_value &= ~(mask << shift); + reg_value |= value << shift; + wm8776_write_cached(chip, reg_index, reg_value); +} + +static int wm8776_field_set(struct snd_kcontrol *ctl, unsigned int value) +{ + struct oxygen *chip = ctl->private_data; + u8 min, max; + int changed; + + min = (ctl->private_value >> 8) & 0xf; + max = (ctl->private_value >> 12) & 0xf; + if (value < min || value > max) + return -EINVAL; + mutex_lock(&chip->mutex); + changed = value != (ctl->private_value & 0xf); + if (changed) { + ctl->private_value = (ctl->private_value & ~0xf) | value; + wm8776_field_set_from_ctl(ctl); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_field_enum_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + value->value.enumerated.item[0] = ctl->private_value & 0xf; + return 0; +} + +static int wm8776_field_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + value->value.integer.value[0] = ctl->private_value & 0xf; + return 0; +} + +static int wm8776_field_enum_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + return wm8776_field_set(ctl, value->value.enumerated.item[0]); +} + +static int wm8776_field_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + return wm8776_field_set(ctl, value->value.integer.value[0]); +} + +static int wm8776_hp_vol_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0x79 - 60; + info->value.integer.max = 0x7f; + return 0; +} + +static int wm8776_hp_vol_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = + data->wm8776_regs[WM8776_HPLVOL] & WM8776_HPATT_MASK; + value->value.integer.value[1] = + data->wm8776_regs[WM8776_HPRVOL] & WM8776_HPATT_MASK; + mutex_unlock(&chip->mutex); + return 0; +} + +static int wm8776_hp_vol_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u8 to_update; + + mutex_lock(&chip->mutex); + to_update = (value->value.integer.value[0] != + (data->wm8776_regs[WM8776_HPLVOL] & WM8776_HPATT_MASK)) + << 0; + to_update |= (value->value.integer.value[1] != + (data->wm8776_regs[WM8776_HPRVOL] & WM8776_HPATT_MASK)) + << 1; + if (value->value.integer.value[0] == value->value.integer.value[1]) { + if (to_update) { + wm8776_write(chip, WM8776_HPMASTER, + value->value.integer.value[0] | + WM8776_HPZCEN | WM8776_UPDATE); + data->wm8776_regs[WM8776_HPLVOL] = + value->value.integer.value[0] | WM8776_HPZCEN; + data->wm8776_regs[WM8776_HPRVOL] = + value->value.integer.value[0] | WM8776_HPZCEN; + } + } else { + if (to_update & 1) + wm8776_write(chip, WM8776_HPLVOL, + value->value.integer.value[0] | + WM8776_HPZCEN | + ((to_update & 2) ? 0 : WM8776_UPDATE)); + if (to_update & 2) + wm8776_write(chip, WM8776_HPRVOL, + value->value.integer.value[1] | + WM8776_HPZCEN | WM8776_UPDATE); + } + mutex_unlock(&chip->mutex); + return to_update != 0; +} + +static int wm8776_input_mux_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mux_bit = ctl->private_value; + + value->value.integer.value[0] = + !!(data->wm8776_regs[WM8776_ADCMUX] & mux_bit); + return 0; +} + +static int wm8776_input_mux_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mux_bit = ctl->private_value; + u16 reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->wm8776_regs[WM8776_ADCMUX]; + if (value->value.integer.value[0]) { + reg &= ~0x003; + reg |= mux_bit; + } else + reg &= ~mux_bit; + changed = reg != data->wm8776_regs[WM8776_ADCMUX]; + if (changed) { + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + reg & 1 ? GPIO_DS_INPUT_ROUTE : 0, + GPIO_DS_INPUT_ROUTE); + wm8776_write(chip, WM8776_ADCMUX, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_input_vol_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0xa5; + info->value.integer.max = 0xff; + return 0; +} + +static int wm8776_input_vol_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = + data->wm8776_regs[WM8776_ADCLVOL] & WM8776_AGMASK; + value->value.integer.value[1] = + data->wm8776_regs[WM8776_ADCRVOL] & WM8776_AGMASK; + mutex_unlock(&chip->mutex); + return 0; +} + +static int wm8776_input_vol_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + int changed = 0; + + mutex_lock(&chip->mutex); + changed = (value->value.integer.value[0] != + (data->wm8776_regs[WM8776_ADCLVOL] & WM8776_AGMASK)) || + (value->value.integer.value[1] != + (data->wm8776_regs[WM8776_ADCRVOL] & WM8776_AGMASK)); + wm8776_write_cached(chip, WM8776_ADCLVOL, + value->value.integer.value[0] | WM8776_ZCA); + wm8776_write_cached(chip, WM8776_ADCRVOL, + value->value.integer.value[1] | WM8776_ZCA); + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_level_control_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "None", "Peak Limiter", "Automatic Level Control" + }; + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int wm8776_level_control_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + if (!(data->wm8776_regs[WM8776_ALCCTRL2] & WM8776_LCEN)) + value->value.enumerated.item[0] = 0; + else if ((data->wm8776_regs[WM8776_ALCCTRL1] & WM8776_LCSEL_MASK) == + WM8776_LCSEL_LIMITER) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + return 0; +} + +static void activate_control(struct oxygen *chip, + struct snd_kcontrol *ctl, unsigned int mode) +{ + unsigned int access; + + if (ctl->private_value & mode) + access = 0; + else + access = SNDRV_CTL_ELEM_ACCESS_INACTIVE; + if ((ctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_INACTIVE) != access) { + ctl->vd[0].access ^= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + } +} + +static int wm8776_level_control_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mode = 0, i; + u16 ctrl1, ctrl2; + int changed; + + if (value->value.enumerated.item[0] >= 3) + return -EINVAL; + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != ctl->private_value; + if (changed) { + ctl->private_value = value->value.enumerated.item[0]; + ctrl1 = data->wm8776_regs[WM8776_ALCCTRL1]; + ctrl2 = data->wm8776_regs[WM8776_ALCCTRL2]; + switch (value->value.enumerated.item[0]) { + default: + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 & ~WM8776_LCEN); + break; + case 1: + wm8776_write_cached(chip, WM8776_ALCCTRL1, + (ctrl1 & ~WM8776_LCSEL_MASK) | + WM8776_LCSEL_LIMITER); + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 | WM8776_LCEN); + mode = LC_CONTROL_LIMITER; + break; + case 2: + wm8776_write_cached(chip, WM8776_ALCCTRL1, + (ctrl1 & ~WM8776_LCSEL_MASK) | + WM8776_LCSEL_ALC_STEREO); + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 | WM8776_LCEN); + mode = LC_CONTROL_ALC; + break; + } + for (i = 0; i < ARRAY_SIZE(data->lc_controls); ++i) + activate_control(chip, data->lc_controls[i], mode); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "None", "High-pass Filter" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + value->value.enumerated.item[0] = + !(data->wm8776_regs[WM8776_ADCIFCTRL] & WM8776_ADCHPD); + return 0; +} + +static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->wm8776_regs[WM8776_ADCIFCTRL] & ~WM8776_ADCHPD; + if (!value->value.enumerated.item[0]) + reg |= WM8776_ADCHPD; + changed = reg != data->wm8776_regs[WM8776_ADCIFCTRL]; + if (changed) + wm8776_write(chip, WM8776_ADCIFCTRL, reg); + mutex_unlock(&chip->mutex); + return changed; +} + +#define WM8776_BIT_SWITCH(xname, reg, bit, invert, flags) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = wm8776_bit_switch_get, \ + .put = wm8776_bit_switch_put, \ + .private_value = ((reg) << 16) | (bit) | ((invert) << 24) | (flags), \ +} +#define _WM8776_FIELD_CTL(xname, reg, shift, initval, min, max, mask, flags) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .private_value = (initval) | ((min) << 8) | ((max) << 12) | \ + ((mask) << 16) | ((shift) << 20) | ((reg) << 24) | (flags) +#define WM8776_FIELD_CTL_ENUM(xname, reg, shift, init, min, max, mask, flags) {\ + _WM8776_FIELD_CTL(xname " Capture Enum", \ + reg, shift, init, min, max, mask, flags), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_INACTIVE, \ + .info = wm8776_field_enum_info, \ + .get = wm8776_field_enum_get, \ + .put = wm8776_field_enum_put, \ +} +#define WM8776_FIELD_CTL_VOLUME(a, b, c, d, e, f, g, h, tlv_p) { \ + _WM8776_FIELD_CTL(a " Capture Volume", b, c, d, e, f, g, h), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_INACTIVE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = wm8776_field_volume_info, \ + .get = wm8776_field_volume_get, \ + .put = wm8776_field_volume_put, \ + .tlv = { .p = tlv_p }, \ +} + +static const DECLARE_TLV_DB_SCALE(wm87x6_dac_db_scale, -6000, 50, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_adc_db_scale, -2100, 50, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_hp_db_scale, -6000, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_lct_db_scale, -1600, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxgain_db_scale, 0, 400, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_ngth_db_scale, -7800, 600, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxatten_lim_db_scale, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxatten_alc_db_scale, -2100, 400, 0); + +static const struct snd_kcontrol_new ds_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .info = wm8776_hp_vol_info, + .get = wm8776_hp_vol_get, + .put = wm8776_hp_vol_put, + .tlv = { .p = wm8776_hp_db_scale }, + }, + WM8776_BIT_SWITCH("Headphone Playback Switch", + WM8776_PWRDOWN, WM8776_HPPD, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Capture Volume", + .info = wm8776_input_vol_info, + .get = wm8776_input_vol_get, + .put = wm8776_input_vol_put, + .tlv = { .p = wm8776_adc_db_scale }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Switch", + .info = snd_ctl_boolean_mono_info, + .get = wm8776_input_mux_get, + .put = wm8776_input_mux_put, + .private_value = 1 << 0, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Capture Switch", + .info = snd_ctl_boolean_mono_info, + .get = wm8776_input_mux_get, + .put = wm8776_input_mux_put, + .private_value = 1 << 1, + }, + WM8776_BIT_SWITCH("Aux", WM8776_ADCMUX, 1 << 2, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Level Control Capture Enum", + .info = wm8776_level_control_info, + .get = wm8776_level_control_get, + .put = wm8776_level_control_put, + .private_value = 0, + }, +}; +static const struct snd_kcontrol_new lc_controls[] = { + WM8776_FIELD_CTL_VOLUME("Limiter Threshold", + WM8776_ALCCTRL1, 0, 11, 0, 15, 0xf, + LC_CONTROL_LIMITER, wm8776_lct_db_scale), + WM8776_FIELD_CTL_ENUM("Limiter Attack Time", + WM8776_ALCCTRL3, 0, 2, 0, 10, 0xf, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_ENUM("Limiter Decay Time", + WM8776_ALCCTRL3, 4, 3, 0, 10, 0xf, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_ENUM("Limiter Transient Window", + WM8776_LIMITER, 4, 2, 0, 7, 0x7, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_VOLUME("Limiter Maximum Attenuation", + WM8776_LIMITER, 0, 6, 3, 12, 0xf, + LC_CONTROL_LIMITER, + wm8776_maxatten_lim_db_scale), + WM8776_FIELD_CTL_VOLUME("ALC Target Level", + WM8776_ALCCTRL1, 0, 11, 0, 15, 0xf, + LC_CONTROL_ALC, wm8776_lct_db_scale), + WM8776_FIELD_CTL_ENUM("ALC Attack Time", + WM8776_ALCCTRL3, 0, 2, 0, 10, 0xf, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_ENUM("ALC Decay Time", + WM8776_ALCCTRL3, 4, 3, 0, 10, 0xf, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_VOLUME("ALC Maximum Gain", + WM8776_ALCCTRL1, 4, 7, 1, 7, 0x7, + LC_CONTROL_ALC, wm8776_maxgain_db_scale), + WM8776_FIELD_CTL_VOLUME("ALC Maximum Attenuation", + WM8776_LIMITER, 0, 10, 10, 15, 0xf, + LC_CONTROL_ALC, wm8776_maxatten_alc_db_scale), + WM8776_FIELD_CTL_ENUM("ALC Hold Time", + WM8776_ALCCTRL2, 0, 0, 0, 15, 0xf, + LC_CONTROL_ALC), + WM8776_BIT_SWITCH("Noise Gate Capture Switch", + WM8776_NOISEGATE, WM8776_NGAT, 0, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_VOLUME("Noise Gate Threshold", + WM8776_NOISEGATE, 2, 0, 0, 7, 0x7, + LC_CONTROL_ALC, wm8776_ngth_db_scale), +}; + +static int xonar_ds_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + return 0; +} + +static int xonar_ds_mixer_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + struct snd_kcontrol *ctl; + int err; + + for (i = 0; i < ARRAY_SIZE(ds_controls); ++i) { + ctl = snd_ctl_new1(&ds_controls[i], chip); + if (!ctl) + return -ENOMEM; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + } + BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); + for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { + ctl = snd_ctl_new1(&lc_controls[i], chip); + if (!ctl) + return -ENOMEM; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + data->lc_controls[i] = ctl; + } + return 0; +} + +static const struct oxygen_model model_xonar_ds = { + .shortname = "Xonar DS", + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_ds_init, + .control_filter = xonar_ds_control_filter, + .mixer_init = xonar_ds_mixer_init, + .cleanup = xonar_ds_cleanup, + .suspend = xonar_ds_suspend, + .resume = xonar_ds_resume, + .pcm_hardware_filter = wm8776_adc_hardware_filter, + .get_i2s_mclk = oxygen_default_i2s_mclk, + .set_dac_params = set_wm87x6_dac_params, + .set_adc_params = set_wm8776_adc_params, + .update_dac_volume = update_wm87x6_volume, + .update_dac_mute = update_wm87x6_mute, + .gpio_changed = xonar_ds_gpio_changed, + .dac_tlv = wm87x6_dac_db_scale, + .model_data_size = sizeof(struct xonar_wm87x6), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1, + .dac_channels = 8, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_SPI, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +int __devinit get_xonar_wm87x6_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + switch (id->subdevice) { + case 0x838e: + chip->model = model_xonar_ds; + break; + default: + return -EINVAL; + } + return 0; +} -- cgit v0.10.2 From a5b5a0649a84db1a0cc1e19997572be8ef3b8c81 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 19 Jan 2010 11:15:45 +0200 Subject: ASoC: tlv320dac33: Correct the prefill number of samples Set the prefill number of samples as the same as the lower threshold in mode7. In this way the codec will read the same amount of data on startup and during the running playback. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 2df9c20..65683aa 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -559,7 +559,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: dac33_write16(codec, DAC33_PREFILL_MSB, - DAC33_THRREG(20)); + DAC33_THRREG(10)); break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", -- cgit v0.10.2 From 84740ac19a0aeb87d1dc21e9d7d517f11bd49748 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Tue, 19 Jan 2010 08:39:05 +0100 Subject: ASoC: fix compile breakage - add a missing header include Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ca24e7f..061f16d 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -16,6 +16,8 @@ #include +#include + struct snd_pcm_substream; /* -- cgit v0.10.2 From 6cd6cede8c33364d8e1abb5ea35adf627e3781b0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 20 Jan 2010 09:39:35 +0200 Subject: ASoC: tlv320dac33: BCLK divider fix The BCLK divider was not configured in case of mode7. This leads to unpredictable behavior when switching between FIFO modes. Configure the BCLK divider depending on the fifo_mode (FIFO is in use, or FIFO bypass). Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 65683aa..e1aa66f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -845,11 +845,14 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); - switch (dac33->fifo_mode) { - case DAC33_FIFO_MODE1: - /* 20: BCLK divide ratio */ + /* BCLK divide ratio */ + if (dac33->fifo_mode) dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + else + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + switch (dac33->fifo_mode) { + case DAC33_FIFO_MODE1: dac33_write16(codec, DAC33_ATHR_MSB, DAC33_THRREG(dac33->alarm_threshold)); break; @@ -864,8 +867,6 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) DAC33_THRREG(10)); break; default: - /* BYPASS mode */ - dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); break; } -- cgit v0.10.2 From 6aceabb459c07a3fb4873c8306de8143c56241b2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 20 Jan 2010 09:39:36 +0200 Subject: ASoC: tlv320dac33: Burst mode BCLK divider configuration Add possibility to configure the burst mode BCLK divider through platform data structure. The BCLK divider changes the actual speed of the serial bus in burst mode, which is faster than the sampling frequency of the running stream. In this way platforms can experiment with the optimal burst speed without the need to modify the codec driver itself. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h index 5858d06..ac06652 100644 --- a/include/sound/tlv320dac33-plat.h +++ b/include/sound/tlv320dac33-plat.h @@ -15,6 +15,7 @@ struct tlv320dac33_platform_data { int power_gpio; + u8 burst_bclkdiv; }; #endif /* __TLV320DAC33_PLAT_H */ diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index e1aa66f..1b35d0c 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -91,6 +91,7 @@ struct tlv320dac33_priv { * this */ enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */ unsigned int nsample; /* burst read amount from host */ + u8 burst_bclkdiv; /* BCLK divider value in burst mode */ enum dac33_state state; }; @@ -845,9 +846,18 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); - /* BCLK divide ratio */ + /* + * BCLK divide ratio + * 0: 1.5 + * 1: 1 + * 2: 2 + * ... + * 254: 254 + * 255: 255 + */ if (dac33->fifo_mode) - dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, + dac33->burst_bclkdiv); else dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); @@ -1239,6 +1249,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, dac33); dac33->power_gpio = pdata->power_gpio; + dac33->burst_bclkdiv = pdata->burst_bclkdiv; dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; /* Disable FIFO use by default */ -- cgit v0.10.2 From c91a988dc6551c66418690e36b2a23cdb0255da8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 21 Jan 2010 10:32:15 +0100 Subject: ALSA: pcm_core: Fix wake_up() optimization This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O" commit. New sleeping queue is introduced to separate user space and kernel space wake_ups. runtime->nowake is renamed to twake (transfer wake). Signed-off-by: Jaroslav Kysela diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e26fb3c..3bc9bca 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -311,8 +311,9 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int nowake: 1; /* no wakeup (data-copy in progress) */ - wait_queue_head_t sleep; + unsigned int twake: 1; /* do transfer (!poll) wakeup */ + wait_queue_head_t sleep; /* poll sleep */ + wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; /* -- private section -- */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index df57a0e..0d428d0 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -894,6 +894,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, memset((void*)runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); + init_waitqueue_head(&runtime->tsleep); runtime->status->state = SNDRV_PCM_STATE_OPEN; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5417f7d..e2a817e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -285,8 +285,8 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (!runtime->nowake && avail >= runtime->control->avail_min) - wake_up(&runtime->sleep); + if (avail >= runtime->control->avail_min) + wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); return 0; } @@ -1692,7 +1692,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, long tout; init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->sleep, &wait); + add_wait_queue(&runtime->tsleep, &wait); for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; @@ -1735,7 +1735,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, break; } _endloop: - remove_wait_queue(&runtime->sleep, &wait); + remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; } @@ -1794,7 +1794,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->nowake = 1; + runtime->twake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1816,7 +1816,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { - runtime->nowake = 0; + runtime->twake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } @@ -1855,7 +1855,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } } _end_unlock: - runtime->nowake = 0; + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); @@ -2016,7 +2016,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->nowake = 1; + runtime->twake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2045,7 +2045,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { - runtime->nowake = 0; + runtime->twake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } @@ -2078,7 +2078,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, xfer += frames; } _end_unlock: - runtime->nowake = 0; + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 27284f6..56ec35e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -919,6 +919,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) runtime->status->state = state; } wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } static struct action_ops snd_pcm_action_stop = { @@ -1004,6 +1005,7 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) SNDRV_TIMER_EVENT_MPAUSE, &runtime->trigger_tstamp); wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; if (substream->timer) @@ -1061,6 +1063,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } static struct action_ops snd_pcm_action_suspend = { -- cgit v0.10.2 From b91b8fa02482a5a18f598ee5d2cd42970051731b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 20 Jan 2010 18:18:35 +0000 Subject: ASoC: Remove console DAPM debug code The same information is now visible via debugfs and with large modern devices dumping everything to the console can be very resource intensive, causing more harm than good. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index de22c2f..d8e9374 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -44,13 +44,6 @@ #include #include -/* debug */ -#ifdef DEBUG -#define dump_dapm(codec, action) dbg_dump_dapm(codec, action) -#else -#define dump_dapm(codec, action) -#endif - /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -1063,66 +1056,6 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return 0; } -#ifdef DEBUG -static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) -{ - struct snd_soc_dapm_widget *w; - struct snd_soc_dapm_path *p = NULL; - int in, out; - - printk("DAPM %s %s\n", codec->name, action); - - list_for_each_entry(w, &codec->dapm_widgets, list) { - - /* only display widgets that effect routing */ - switch (w->id) { - case snd_soc_dapm_pre: - case snd_soc_dapm_post: - case snd_soc_dapm_vmid: - continue; - case snd_soc_dapm_mux: - case snd_soc_dapm_value_mux: - case snd_soc_dapm_output: - case snd_soc_dapm_input: - case snd_soc_dapm_switch: - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: - case snd_soc_dapm_spk: - case snd_soc_dapm_line: - case snd_soc_dapm_micbias: - case snd_soc_dapm_dac: - case snd_soc_dapm_adc: - case snd_soc_dapm_pga: - case snd_soc_dapm_mixer: - case snd_soc_dapm_mixer_named_ctl: - case snd_soc_dapm_supply: - case snd_soc_dapm_aif_in: - case snd_soc_dapm_aif_out: - if (w->name) { - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - printk("%s: %s in %d out %d\n", w->name, - w->power ? "On":"Off",in, out); - - list_for_each_entry(p, &w->sources, list_sink) { - if (p->connect) - printk(" in %s %s\n", p->name ? p->name : "static", - p->source->name); - } - list_for_each_entry(p, &w->sinks, list_source) { - if (p->connect) - printk(" out %s %s\n", p->name ? p->name : "static", - p->sink->name); - } - } - break; - } - } -} -#endif - #ifdef CONFIG_DEBUG_FS static int dapm_widget_power_open_file(struct inode *inode, struct file *file) { @@ -1254,10 +1187,8 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, path->connect = 0; /* old connection must be powered down */ } - if (found) { + if (found) dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(widget->codec, "mux power update"); - } return 0; } @@ -1285,10 +1216,8 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, break; } - if (found) { + if (found) dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(widget->codec, "mixer power update"); - } return 0; } @@ -1404,9 +1333,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, */ int snd_soc_dapm_sync(struct snd_soc_codec *codec) { - int ret = dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(codec, "sync"); - return ret; + return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); @@ -2163,7 +2090,6 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, dapm_power_widgets(codec, event); mutex_unlock(&codec->mutex); - dump_dapm(codec, __func__); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); -- cgit v0.10.2 From a96ca3387382498ec8b501db5acef3ed9eb1bd36 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 19 Jan 2010 22:49:43 +0000 Subject: ASoC: Support turning off bias when the CODEC is idle Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..a8768ea 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -405,6 +405,8 @@ struct snd_soc_codec { short reg_cache_size; short reg_cache_step; + unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + /* dapm */ u32 pop_time; struct list_head dapm_widgets; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d8e9374..6c33510 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1012,13 +1012,28 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) sys_power = 0; break; case SND_SOC_DAPM_STREAM_NOP: - sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + sys_power = 0; + break; + default: + sys_power = 1; + break; + } break; default: break; } } + if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); + if (ret != 0) + pr_err("Failed to turn on bias: %d\n", ret); + } + /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { @@ -1042,6 +1057,14 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) pr_err("Failed to apply standby bias: %d\n", ret); } + /* If we're in standby and can support bias off then do that */ + if (codec->bias_level == SND_SOC_BIAS_STANDBY && + codec->idle_bias_off) { + ret = snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + if (ret != 0) + pr_err("Failed to turn off bias: %d\n", ret); + } + /* If we just powered up then move to active bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { ret = snd_soc_dapm_set_bias_level(socdev, -- cgit v0.10.2 From 821dd91ec7838e1313d783384ea9ce43510d4013 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 21 Jan 2010 11:33:20 +0000 Subject: ASoC: Use BIAS_OFF when idle for wm_hubs devices This provides a small power saving when audio is inactive. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index d73c305..a67319d 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -753,6 +753,12 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, WM8993_LINEOUT2_MODE, WM8993_LINEOUT2_MODE); + /* If the line outputs are differential then we aren't presenting + * VMID as an output and can disable it. + */ + if (lineout1_diff && lineout2_diff) + codec->idle_bias_off = 1; + if (lineout1fb) snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); -- cgit v0.10.2 From fd0b092a7b14559e2ff17ef3aaefb5d8adc7e15f Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 21 Jan 2010 14:54:38 +0100 Subject: ALSA: hda - AD1988 codec - fix SPDIF-input mixer initialization (unmute) The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate pin to get captured samples instead zeros. Tested on Lenovo Thinkstation. Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cecd3c1..865715e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2458,6 +2458,12 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { { } }; +static struct hda_verb ad1988_spdif_in_init_verbs[] = { + /* unmute SPDIF input pin */ + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + /* AD1989 has no ADC -> SPDIF route */ static struct hda_verb ad1989_spdif_init_verbs[] = { /* SPDIF-1 out pin */ @@ -3193,8 +3199,11 @@ static int patch_ad1988(struct hda_codec *codec) ad1988_spdif_init_verbs; } } - if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) + if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) { spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1988_spdif_in_init_verbs; + } codec->patch_ops = ad198x_patch_ops; switch (board_config) { -- cgit v0.10.2 From 5f6c3de6a79820de124fa2bb1b77d43a09410e42 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 23 Jan 2010 22:19:29 +0100 Subject: ALSA: hda - Minor fixes for Compaq Presario F700 quirk Minor fixes for HP Compaq Presario F700 quirks with Cxt5051 codec: - changed the capture mixer elements to the standard name. - fixed the quirk name string without a space - sorted the quirk list - updated the documentation Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index e72cee9..cb46eb2 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -279,6 +279,7 @@ Conexant 5051 laptop Basic Laptop config (default) hp HP Spartan laptop hp-dv6736 HP dv6736 + hp-f700 HP Compaq Presario F700 lenovo-x200 Lenovo X200 laptop Conexant 5066 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 685015a..084600e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1742,8 +1742,8 @@ static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { }; static struct snd_kcontrol_new cxt5051_f700_mixers[] = { - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1901,17 +1901,17 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", - [CXT5051_F700] = "hp 700" + [CXT5051_F700] = "hp-700", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), + SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), - SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), {} }; -- cgit v0.10.2 From 4e4ac60030293cb3d1e4bacf7c8be9aebdb8df61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 23 Jan 2010 22:29:54 +0100 Subject: ALSA: hda - Fix HP dv6736 capture mixer name Use the standard "Capture" mixer name for HP dv6736 with Cxt5051 codec. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 084600e..08c5b32 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1726,8 +1726,8 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { }; static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, -- cgit v0.10.2 From faddaa5d1c0cd29629c9c7e7a9d41ecb3149a064 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 23 Jan 2010 22:31:36 +0100 Subject: ALSA: hda - Add support for Toshiba Satellite M300 Added the support for Toshiba Satellite M300 with Conexant 5051 codec. Since the laptop has no port C connection and the pin reports always the jack sense true, we need to ignore port-C unsol event. Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index cb46eb2..8f06f20 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -281,6 +281,7 @@ Conexant 5051 hp-dv6736 HP dv6736 hp-f700 HP Compaq Presario F700 lenovo-x200 Lenovo X200 laptop + toshiba Toshiba Satellite M300 Conexant 5066 ============= diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 08c5b32..56dda9c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -46,6 +46,8 @@ #define CXT5051_PORTB_EVENT 0x38 #define CXT5051_PORTC_EVENT 0x39 +#define AUTO_MIC_PORTB (1 << 1) +#define AUTO_MIC_PORTC (1 << 2) struct conexant_jack { @@ -74,7 +76,7 @@ struct conexant_spec { */ unsigned int cur_eapd; unsigned int hp_present; - unsigned int no_auto_mic; + unsigned int auto_mic; unsigned int need_dac_fix; /* capture */ @@ -1626,7 +1628,7 @@ static void cxt5051_portb_automic(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int present; - if (spec->no_auto_mic) + if (!(spec->auto_mic & AUTO_MIC_PORTB)) return; present = snd_hda_jack_detect(codec, 0x17); snd_hda_codec_write(codec, 0x14, 0, @@ -1641,7 +1643,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec) unsigned int present; hda_nid_t new_adc; - if (spec->no_auto_mic) + if (!(spec->auto_mic & AUTO_MIC_PORTC)) return; present = snd_hda_jack_detect(codec, 0x18); if (present) @@ -1757,6 +1759,24 @@ static struct snd_kcontrol_new cxt5051_f700_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { + HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1893,6 +1913,7 @@ enum { CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_F700, /* HP Compaq Presario F700 */ + CXT5051_TOSHIBA, /* Toshiba M300 & co */ CXT5051_MODELS }; @@ -1902,12 +1923,14 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", [CXT5051_F700] = "hp-700", + [CXT5051_TOSHIBA] = "toshiba", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), + SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba M30x", CXT5051_TOSHIBA), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), @@ -1950,6 +1973,7 @@ static int patch_cxt5051(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, cxt5051_models, cxt5051_cfg_tbl); + spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC; switch (board_config) { case CXT5051_HP: spec->mixers[0] = cxt5051_hp_mixers; @@ -1957,7 +1981,7 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_HP_DV6736: spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; spec->mixers[0] = cxt5051_hp_dv6736_mixers; - spec->no_auto_mic = 1; + spec->auto_mic = 0; break; case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; @@ -1965,7 +1989,11 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_F700: spec->init_verbs[0] = cxt5051_f700_init_verbs; spec->mixers[0] = cxt5051_f700_mixers; - spec->no_auto_mic = 1; + spec->auto_mic = 0; + break; + case CXT5051_TOSHIBA: + spec->mixers[0] = cxt5051_toshiba_mixers; + spec->auto_mic = AUTO_MIC_PORTB; break; } -- cgit v0.10.2 From 2c7a3fb3f81df7318c70d2b8ecbd87f008e28d52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Jan 2010 10:47:02 +0100 Subject: ALSA: hda - Merge playback controls for Cx5051 codec models All cx5051 codec models have the same Master playback mixer definitions. Merge them together. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56dda9c..e24bec6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1689,13 +1689,7 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec, conexant_report_jack(codec, nid); } -static struct snd_kcontrol_new cxt5051_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), +static struct snd_kcontrol_new cxt5051_playback_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1705,7 +1699,16 @@ static struct snd_kcontrol_new cxt5051_mixers[] = { .put = cxt5051_hp_master_sw_put, .private_value = 0x1a, }, + {} +}; +static struct snd_kcontrol_new cxt5051_capture_mixers[] = { + HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), {} }; @@ -1714,48 +1717,18 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("External Mic Volume", 0x15, 0x00, HDA_INPUT), HDA_CODEC_MUTE("External Mic Switch", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} }; static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} }; static struct snd_kcontrol_new cxt5051_f700_mixers[] = { HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} }; @@ -1764,16 +1737,6 @@ static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} }; @@ -1958,8 +1921,9 @@ static int patch_cxt5051(struct hda_codec *codec) spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT; spec->num_adc_nids = 1; /* not 2; via auto-mic switch */ spec->adc_nids = cxt5051_adc_nids; - spec->num_mixers = 1; - spec->mixers[0] = cxt5051_mixers; + spec->num_mixers = 2; + spec->mixers[0] = cxt5051_capture_mixers; + spec->mixers[1] = cxt5051_playback_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5051_init_verbs; spec->spdif_route = 0; -- cgit v0.10.2 From 6953e5524a2ee0dcf57a83d8a6728d1262c54c37 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Jan 2010 11:00:27 +0100 Subject: ALSA: hda - initialize mic port on cxt5051 codec dynamically Initialize the mic ports B & C on Conexant 5051 codec dynamically according to the mic jack detection, instead of static init arrays. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e24bec6..4fbb398 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1765,8 +1765,6 @@ static struct hda_verb cxt5051_init_verbs[] = { /* EAPD */ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, { } /* end */ }; @@ -1792,7 +1790,6 @@ static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { /* EAPD */ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, { } /* end */ }; @@ -1824,8 +1821,6 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { /* EAPD */ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, { } /* end */ }; @@ -1852,15 +1847,34 @@ static struct hda_verb cxt5051_f700_init_verbs[] = { /* EAPD */ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, { } /* end */ }; +static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, + unsigned int event) +{ + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | event); +#ifdef CONFIG_SND_HDA_INPUT_JACK + conexant_add_jack(codec, nid, SND_JACK_MICROPHONE); + conexant_report_jack(codec, nid); +#endif +} + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + conexant_init(codec); conexant_init_jacks(codec); + + if (spec->auto_mic & AUTO_MIC_PORTB) + cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT); + if (spec->auto_mic & AUTO_MIC_PORTC) + cxt5051_init_mic_port(codec, 0x18, CXT5051_PORTC_EVENT); + if (codec->patch_ops.unsol_event) { cxt5051_hp_automute(codec); cxt5051_portb_automic(codec); -- cgit v0.10.2 From ecda0cff9df77d3f7d388bd4966e61f1947d2c95 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Jan 2010 11:14:36 +0100 Subject: ALSA: hda - Fix SPDIF output widget for Cxt5051 codec Fixed the wrongly set up for SPDIF output on Conexant 5051 codec. It must point to the audio out widget instead of a pin. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4fbb398..250b74f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -42,7 +42,7 @@ /* Conexant 5051 specific */ -#define CXT5051_SPDIF_OUT 0x1C +#define CXT5051_SPDIF_OUT 0x12 #define CXT5051_PORTB_EVENT 0x38 #define CXT5051_PORTC_EVENT 0x39 -- cgit v0.10.2 From 23d2df5b0db67fa90d3caf4b2d2f21ca33ec9c11 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Jan 2010 11:19:27 +0100 Subject: ALSA: hda - Change headphone pin control with master volume on cx5051 The HP pin (0x16) control has to be changed dynamically depending on the master volume switch as well as the speaker pin (0x1a). Otherwise the headphone still sounds with master off. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 250b74f..9077e41 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1605,6 +1605,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; unsigned int pinctl; + /* headphone pin */ + pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0; + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + /* speaker pin */ pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); -- cgit v0.10.2 From 95f475f7a2e5d60fe9eeb7a2700753036a6ee6a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jan 2010 15:41:11 +0100 Subject: ALSA: hda - Remove coef output in Realtek proc files The output of COEF index/value in the proc file for Realtek codecs is rather useless since the value varies together with the index. Let's get rid of it again. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c53faa9..a3d2238 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -841,27 +841,6 @@ static void add_verb(struct alc_spec *spec, const struct hda_verb *verb) spec->init_verbs[spec->num_init_verbs++] = verb; } -#ifdef CONFIG_PROC_FS -/* - * hook for proc - */ -static void print_realtek_coef(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) -{ - int coeff; - - if (nid != 0x20) - return; - coeff = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0); - snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff); - coeff = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_COEF_INDEX, 0); - snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff); -} -#else -#define print_realtek_coef NULL -#endif - /* * set up from the preset table */ @@ -5078,7 +5057,6 @@ static int patch_alc880(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc880_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -6688,7 +6666,6 @@ static int patch_alc260(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc260_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -10306,7 +10283,6 @@ static int patch_alc882(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc882_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -12170,7 +12146,6 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc262_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -13237,8 +13212,6 @@ static int patch_alc268(struct hda_codec *codec) if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; - codec->proc_widget_hook = print_realtek_coef; - return 0; } @@ -13955,7 +13928,6 @@ static int patch_alc269(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc269_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -15083,7 +15055,6 @@ static int patch_alc861(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -16063,7 +16034,6 @@ static int patch_alc861vd(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc861vd_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } @@ -18198,7 +18168,6 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc662_loopbacks; #endif - codec->proc_widget_hook = print_realtek_coef; return 0; } -- cgit v0.10.2 From 0aea778efa0d632b62eb35122cbb3b9fae548c61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Jan 2010 15:44:11 +0100 Subject: ALSA: hda - Remove the COEF setup for ALC267/ALC268 The COEF setup for model=auto seems problematic on some laptops, resulting in the silent speaker output. Better to disable it for now. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a3d2238..b2f543d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1145,6 +1145,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0888: alc888_coef_init(codec); break; +#if 0 /* XXX: This may cause the silent output on speaker on some machines */ case 0x10ec0267: case 0x10ec0268: snd_hda_codec_write(codec, 0x20, 0, @@ -1157,6 +1158,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) AC_VERB_SET_PROC_COEF, tmp | 0x3000); break; +#endif /* XXX */ } break; } -- cgit v0.10.2 From 8484c63f4b363d79febe35f95328e38018b65026 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 21 Jan 2010 21:10:47 +0100 Subject: ASoC: add simplified versions of widget macros Many macros from include/sound/soc-dapm.h take an array and a number of elements in it as arguments, whereas most users use static arrays and use "x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of those macros, calling ARRAY_SIZE() internally. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c5c95e1d..c0922a0 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -95,6 +95,21 @@ .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ .num_kcontrols = 1} +/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ +#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\ + wcontrols) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} +#define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols)\ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} +#define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols)\ +{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = ARRAY_SIZE(wcontrols)} + /* path domain with event - event handler must return 0 for success */ #define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \ wncontrols, wevent, wflags) \ @@ -126,6 +141,23 @@ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} +/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ +#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ + .event = wevent, .event_flags = wflags} +#define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ + .event = wevent, .event_flags = wflags} +#define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols, wevent, wflags) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags} + /* events that are pre and post DAPM */ #define SND_SOC_DAPM_PRE(wname, wevent) \ { .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \ -- cgit v0.10.2 From 6c2fb6a8d8c43544e7665859f29373c98d17df75 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 21 Jan 2010 22:04:03 +0100 Subject: ASoC: add helper macros to declare struct soc_enum instances Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index a8768ea..4bbeb9f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -169,6 +169,23 @@ .private_value = (unsigned long)&xenum } /* + * Simplified versions of above macros, declaring a struct and calculating + * ARRAY_SIZE internally + */ +#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \ + struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ + ARRAY_SIZE(xtexts), xtexts) +#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \ + SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) +#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \ + struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts) +#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ + struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations. -- cgit v0.10.2 From 895d4509d069f0706427ca75fcf0929ed136d0d7 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 22 Jan 2010 19:09:03 +0100 Subject: ASoC: add DAI and platform / DMA drivers for SH SIU Several SuperH platforms, including sh7722, sh7343, sh7354, sh7367 include a Sound Interface Unit (SIU). This patch adds DAI and platform / DMA drivers for this interface. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/arch/sh/include/asm/siu.h b/arch/sh/include/asm/siu.h new file mode 100644 index 0000000..57565a3 --- /dev/null +++ b/arch/sh/include/asm/siu.h @@ -0,0 +1,26 @@ +/* + * platform header for the SIU ASoC driver + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef ASM_SIU_H +#define ASM_SIU_H + +#include + +struct device; + +struct siu_platform { + struct device *dma_dev; + enum sh_dmae_slave_chan_id dma_slave_tx_a; + enum sh_dmae_slave_chan_id dma_slave_rx_a; + enum sh_dmae_slave_chan_id dma_slave_tx_b; + enum sh_dmae_slave_chan_id dma_slave_rx_b; +}; + +#endif /* ASM_SIU_H */ diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 8072a6d..3f1cd55 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -26,6 +26,12 @@ config SND_SOC_SH4_FSI help This option enables FSI sound support +config SND_SOC_SH4_SIU + tristate + depends on (SUPERH || ARCH_SHMOBILE) && HAVE_CLK + select DMADEVICES + select SH_DMAE + ## ## Boards ## diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 1d0ec0a..5a97d25 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -6,9 +6,11 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o snd-soc-hac-objs := hac.o snd-soc-ssi-objs := ssi.o snd-soc-fsi-objs := fsi.o +snd-soc-siu-objs := siu_pcm.o siu_dai.o obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o +obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h new file mode 100644 index 0000000..9cc04ab --- /dev/null +++ b/sound/soc/sh/siu.h @@ -0,0 +1,193 @@ +/* + * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * Copyright (C) 2006 Carlos Munoz + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef SIU_H +#define SIU_H + +/* Common kernel and user-space firmware-building defines and types */ + +#define YRAM0_SIZE (0x0040 / 4) /* 16 */ +#define YRAM1_SIZE (0x0080 / 4) /* 32 */ +#define YRAM2_SIZE (0x0040 / 4) /* 16 */ +#define YRAM3_SIZE (0x0080 / 4) /* 32 */ +#define YRAM4_SIZE (0x0080 / 4) /* 32 */ +#define YRAM_DEF_SIZE (YRAM0_SIZE + YRAM1_SIZE + YRAM2_SIZE + \ + YRAM3_SIZE + YRAM4_SIZE) +#define YRAM_FIR_SIZE (0x0400 / 4) /* 256 */ +#define YRAM_IIR_SIZE (0x0200 / 4) /* 128 */ + +#define XRAM0_SIZE (0x0400 / 4) /* 256 */ +#define XRAM1_SIZE (0x0200 / 4) /* 128 */ +#define XRAM2_SIZE (0x0200 / 4) /* 128 */ + +/* PRAM program array size */ +#define PRAM0_SIZE (0x0100 / 4) /* 64 */ +#define PRAM1_SIZE ((0x2000 - 0x0100) / 4) /* 1984 */ + +#include + +struct siu_spb_param { + __u32 ab1a; /* input FIFO address */ + __u32 ab0a; /* output FIFO address */ + __u32 dir; /* 0=the ather except CPUOUTPUT, 1=CPUINPUT */ + __u32 event; /* SPB program starting conditions */ + __u32 stfifo; /* STFIFO register setting value */ + __u32 trdat; /* TRDAT register setting value */ +}; + +struct siu_firmware { + __u32 yram_fir_coeff[YRAM_FIR_SIZE]; + __u32 pram0[PRAM0_SIZE]; + __u32 pram1[PRAM1_SIZE]; + __u32 yram0[YRAM0_SIZE]; + __u32 yram1[YRAM1_SIZE]; + __u32 yram2[YRAM2_SIZE]; + __u32 yram3[YRAM3_SIZE]; + __u32 yram4[YRAM4_SIZE]; + __u32 spbpar_num; + struct siu_spb_param spbpar[32]; +}; + +#ifdef __KERNEL__ + +#include +#include +#include + +#include + +#include +#include +#include + +#define SIU_PERIOD_BYTES_MAX 8192 /* DMA transfer/period size */ +#define SIU_PERIOD_BYTES_MIN 256 /* DMA transfer/period size */ +#define SIU_PERIODS_MAX 64 /* Max periods in buffer */ +#define SIU_PERIODS_MIN 4 /* Min periods in buffer */ +#define SIU_BUFFER_BYTES_MAX (SIU_PERIOD_BYTES_MAX * SIU_PERIODS_MAX) + +/* SIU ports: only one can be used at a time */ +enum { + SIU_PORT_A, + SIU_PORT_B, + SIU_PORT_NUM, +}; + +/* SIU clock configuration */ +enum { + SIU_CLKA_PLL, + SIU_CLKA_EXT, + SIU_CLKB_PLL, + SIU_CLKB_EXT +}; + +struct siu_info { + int port_id; + u32 __iomem *pram; + u32 __iomem *xram; + u32 __iomem *yram; + u32 __iomem *reg; + struct siu_firmware fw; +}; + +struct siu_stream { + struct tasklet_struct tasklet; + struct snd_pcm_substream *substream; + snd_pcm_format_t format; + size_t buf_bytes; + size_t period_bytes; + int cur_period; /* Period currently in dma */ + u32 volume; + snd_pcm_sframes_t xfer_cnt; /* Number of frames */ + u8 rw_flg; /* transfer status */ + /* DMA status */ + struct dma_chan *chan; /* DMA channel */ + struct dma_async_tx_descriptor *tx_desc; + dma_cookie_t cookie; + struct sh_dmae_slave param; +}; + +struct siu_port { + unsigned long play_cap; /* Used to track full duplex */ + struct snd_pcm *pcm; + struct siu_stream playback; + struct siu_stream capture; + u32 stfifo; /* STFIFO value from firmware */ + u32 trdat; /* TRDAT value from firmware */ +}; + +extern struct siu_port *siu_ports[SIU_PORT_NUM]; + +static inline struct siu_port *siu_port_info(struct snd_pcm_substream *substream) +{ + struct platform_device *pdev = + to_platform_device(substream->pcm->card->dev); + return siu_ports[pdev->id]; +} + +/* Register access */ +static inline void siu_write32(u32 __iomem *addr, u32 val) +{ + __raw_writel(val, addr); +} + +static inline u32 siu_read32(u32 __iomem *addr) +{ + return __raw_readl(addr); +} + +/* SIU registers */ +#define SIU_IFCTL (0x000 / sizeof(u32)) +#define SIU_SRCTL (0x004 / sizeof(u32)) +#define SIU_SFORM (0x008 / sizeof(u32)) +#define SIU_CKCTL (0x00c / sizeof(u32)) +#define SIU_TRDAT (0x010 / sizeof(u32)) +#define SIU_STFIFO (0x014 / sizeof(u32)) +#define SIU_DPAK (0x01c / sizeof(u32)) +#define SIU_CKREV (0x020 / sizeof(u32)) +#define SIU_EVNTC (0x028 / sizeof(u32)) +#define SIU_SBCTL (0x040 / sizeof(u32)) +#define SIU_SBPSET (0x044 / sizeof(u32)) +#define SIU_SBFSTS (0x068 / sizeof(u32)) +#define SIU_SBDVCA (0x06c / sizeof(u32)) +#define SIU_SBDVCB (0x070 / sizeof(u32)) +#define SIU_SBACTIV (0x074 / sizeof(u32)) +#define SIU_DMAIA (0x090 / sizeof(u32)) +#define SIU_DMAIB (0x094 / sizeof(u32)) +#define SIU_DMAOA (0x098 / sizeof(u32)) +#define SIU_DMAOB (0x09c / sizeof(u32)) +#define SIU_DMAML (0x0a0 / sizeof(u32)) +#define SIU_SPSTS (0x0cc / sizeof(u32)) +#define SIU_SPCTL (0x0d0 / sizeof(u32)) +#define SIU_BRGASEL (0x100 / sizeof(u32)) +#define SIU_BRRA (0x104 / sizeof(u32)) +#define SIU_BRGBSEL (0x108 / sizeof(u32)) +#define SIU_BRRB (0x10c / sizeof(u32)) + +extern struct snd_soc_platform siu_platform; +extern struct snd_soc_dai siu_i2s_dai; + +int siu_init_port(int port, struct siu_port **port_info, struct snd_card *card); +void siu_free_port(struct siu_port *port_info); + +#endif + +#endif /* SIU_H */ diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c new file mode 100644 index 0000000..5452d19 --- /dev/null +++ b/sound/soc/sh/siu_dai.c @@ -0,0 +1,847 @@ +/* + * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * Copyright (C) 2006 Carlos Munoz + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include +#include +#include + +#include +#include + +#include +#include + +#include "siu.h" + +/* Board specifics */ +#if defined(CONFIG_CPU_SUBTYPE_SH7722) +# define SIU_MAX_VOLUME 0x1000 +#else +# define SIU_MAX_VOLUME 0x7fff +#endif + +#define PRAM_SIZE 0x2000 +#define XRAM_SIZE 0x800 +#define YRAM_SIZE 0x800 + +#define XRAM_OFFSET 0x4000 +#define YRAM_OFFSET 0x6000 +#define REG_OFFSET 0xc000 + +#define PLAYBACK_ENABLED 1 +#define CAPTURE_ENABLED 2 + +#define VOLUME_CAPTURE 0 +#define VOLUME_PLAYBACK 1 +#define DFLT_VOLUME_LEVEL 0x08000800 + +/* + * SPDIF is only available on port A and on some SIU implementations it is only + * available for input. Due to the lack of hardware to test it, SPDIF is left + * disabled in this driver version + */ +struct format_flag { + u32 i2s; + u32 pcm; + u32 spdif; + u32 mask; +}; + +struct port_flag { + struct format_flag playback; + struct format_flag capture; +}; + +static struct port_flag siu_flags[SIU_PORT_NUM] = { + [SIU_PORT_A] = { + .playback = { + .i2s = 0x50000000, + .pcm = 0x40000000, + .spdif = 0x80000000, /* not on all SIU versions */ + .mask = 0xd0000000, + }, + .capture = { + .i2s = 0x05000000, + .pcm = 0x04000000, + .spdif = 0x08000000, + .mask = 0x0d000000, + }, + }, + [SIU_PORT_B] = { + .playback = { + .i2s = 0x00500000, + .pcm = 0x00400000, + .spdif = 0, /* impossible - turn off */ + .mask = 0x00500000, + }, + .capture = { + .i2s = 0x00050000, + .pcm = 0x00040000, + .spdif = 0, /* impossible - turn off */ + .mask = 0x00050000, + }, + }, +}; + +static void siu_dai_start(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + + dev_dbg(port_info->pcm->card->dev, "%s\n", __func__); + + /* Turn on SIU clock */ + pm_runtime_get_sync(siu_i2s_dai.dev); + + /* Issue software reset to siu */ + siu_write32(base + SIU_SRCTL, 0); + + /* Wait for the reset to take effect */ + udelay(1); + + port_info->stfifo = 0; + port_info->trdat = 0; + + /* portA, portB, SIU operate */ + siu_write32(base + SIU_SRCTL, 0x301); + + /* portA=256fs, portB=256fs */ + siu_write32(base + SIU_CKCTL, 0x40400000); + + /* portA's BRG does not divide SIUCKA */ + siu_write32(base + SIU_BRGASEL, 0); + siu_write32(base + SIU_BRRA, 0); + + /* portB's BRG divides SIUCKB by half */ + siu_write32(base + SIU_BRGBSEL, 1); + siu_write32(base + SIU_BRRB, 0); + + siu_write32(base + SIU_IFCTL, 0x44440000); + + /* portA: 32 bit/fs, master; portB: 32 bit/fs, master */ + siu_write32(base + SIU_SFORM, 0x0c0c0000); + + /* + * Volume levels: looks like the DSP firmware implements volume controls + * differently from what's described in the datasheet + */ + siu_write32(base + SIU_SBDVCA, port_info->playback.volume); + siu_write32(base + SIU_SBDVCB, port_info->capture.volume); +} + +static void siu_dai_stop(void) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + + /* SIU software reset */ + siu_write32(base + SIU_SRCTL, 0); + + /* Turn off SIU clock */ + pm_runtime_put_sync(siu_i2s_dai.dev); +} + +static void siu_dai_spbAselect(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_firmware *fw = &info->fw; + u32 *ydef = fw->yram0; + u32 idx; + + /* path A use */ + if (!info->port_id) + idx = 1; /* portA */ + else + idx = 2; /* portB */ + + ydef[0] = (fw->spbpar[idx].ab1a << 16) | + (fw->spbpar[idx].ab0a << 8) | + (fw->spbpar[idx].dir << 7) | 3; + ydef[1] = fw->yram0[1]; /* 0x03000300 */ + ydef[2] = (16 / 2) << 24; + ydef[3] = fw->yram0[3]; /* 0 */ + ydef[4] = fw->yram0[4]; /* 0 */ + ydef[7] = fw->spbpar[idx].event; + port_info->stfifo |= fw->spbpar[idx].stfifo; + port_info->trdat |= fw->spbpar[idx].trdat; +} + +static void siu_dai_spbBselect(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_firmware *fw = &info->fw; + u32 *ydef = fw->yram0; + u32 idx; + + /* path B use */ + if (!info->port_id) + idx = 7; /* portA */ + else + idx = 8; /* portB */ + + ydef[5] = (fw->spbpar[idx].ab1a << 16) | + (fw->spbpar[idx].ab0a << 8) | 1; + ydef[6] = fw->spbpar[idx].event; + port_info->stfifo |= fw->spbpar[idx].stfifo; + port_info->trdat |= fw->spbpar[idx].trdat; +} + +static void siu_dai_open(struct siu_stream *siu_stream) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + u32 srctl, ifctl; + + srctl = siu_read32(base + SIU_SRCTL); + ifctl = siu_read32(base + SIU_IFCTL); + + switch (info->port_id) { + case SIU_PORT_A: + /* portA operates */ + srctl |= 0x200; + ifctl &= ~0xc2; + break; + case SIU_PORT_B: + /* portB operates */ + srctl |= 0x100; + ifctl &= ~0x31; + break; + } + + siu_write32(base + SIU_SRCTL, srctl); + /* Unmute and configure portA */ + siu_write32(base + SIU_IFCTL, ifctl); +} + +/* + * At the moment only fixed Left-upper, Left-lower, Right-upper, Right-lower + * packing is supported + */ +static void siu_dai_pcmdatapack(struct siu_stream *siu_stream) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + u32 dpak; + + dpak = siu_read32(base + SIU_DPAK); + + switch (info->port_id) { + case SIU_PORT_A: + dpak &= ~0xc0000000; + break; + case SIU_PORT_B: + dpak &= ~0x00c00000; + break; + } + + siu_write32(base + SIU_DPAK, dpak); +} + +static int siu_dai_spbstart(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_firmware *fw = &info->fw; + u32 *ydef = fw->yram0; + int cnt; + u32 __iomem *add; + u32 *ptr; + + /* Load SPB Program in PRAM */ + ptr = fw->pram0; + add = info->pram; + for (cnt = 0; cnt < PRAM0_SIZE; cnt++, add++, ptr++) + siu_write32(add, *ptr); + + ptr = fw->pram1; + add = info->pram + (0x0100 / sizeof(u32)); + for (cnt = 0; cnt < PRAM1_SIZE; cnt++, add++, ptr++) + siu_write32(add, *ptr); + + /* XRAM initialization */ + add = info->xram; + for (cnt = 0; cnt < XRAM0_SIZE + XRAM1_SIZE + XRAM2_SIZE; cnt++, add++) + siu_write32(add, 0); + + /* YRAM variable area initialization */ + add = info->yram; + for (cnt = 0; cnt < YRAM_DEF_SIZE; cnt++, add++) + siu_write32(add, ydef[cnt]); + + /* YRAM FIR coefficient area initialization */ + add = info->yram + (0x0200 / sizeof(u32)); + for (cnt = 0; cnt < YRAM_FIR_SIZE; cnt++, add++) + siu_write32(add, fw->yram_fir_coeff[cnt]); + + /* YRAM IIR coefficient area initialization */ + add = info->yram + (0x0600 / sizeof(u32)); + for (cnt = 0; cnt < YRAM_IIR_SIZE; cnt++, add++) + siu_write32(add, 0); + + siu_write32(base + SIU_TRDAT, port_info->trdat); + port_info->trdat = 0x0; + + + /* SPB start condition: software */ + siu_write32(base + SIU_SBACTIV, 0); + /* Start SPB */ + siu_write32(base + SIU_SBCTL, 0xc0000000); + /* Wait for program to halt */ + cnt = 0x10000; + while (--cnt && siu_read32(base + SIU_SBCTL) != 0x80000000) + cpu_relax(); + + if (!cnt) + return -EBUSY; + + /* SPB program start address setting */ + siu_write32(base + SIU_SBPSET, 0x00400000); + /* SPB hardware start(FIFOCTL source) */ + siu_write32(base + SIU_SBACTIV, 0xc0000000); + + return 0; +} + +static void siu_dai_spbstop(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + + siu_write32(base + SIU_SBACTIV, 0); + /* SPB stop */ + siu_write32(base + SIU_SBCTL, 0); + + port_info->stfifo = 0; +} + +/* API functions */ + +/* Playback and capture hardware properties are identical */ +static struct snd_pcm_hardware siu_dai_pcm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = SIU_BUFFER_BYTES_MAX, + .period_bytes_min = SIU_PERIOD_BYTES_MIN, + .period_bytes_max = SIU_PERIOD_BYTES_MAX, + .periods_min = SIU_PERIODS_MIN, + .periods_max = SIU_PERIODS_MAX, +}; + +static int siu_dai_info_volume(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_info *uinfo) +{ + struct siu_port *port_info = snd_kcontrol_chip(kctrl); + + dev_dbg(port_info->pcm->card->dev, "%s\n", __func__); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SIU_MAX_VOLUME; + + return 0; +} + +static int siu_dai_get_volume(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) +{ + struct siu_port *port_info = snd_kcontrol_chip(kctrl); + struct device *dev = port_info->pcm->card->dev; + u32 vol; + + dev_dbg(dev, "%s\n", __func__); + + switch (kctrl->private_value) { + case VOLUME_PLAYBACK: + /* Playback is always on port 0 */ + vol = port_info->playback.volume; + ucontrol->value.integer.value[0] = vol & 0xffff; + ucontrol->value.integer.value[1] = vol >> 16 & 0xffff; + break; + case VOLUME_CAPTURE: + /* Capture is always on port 1 */ + vol = port_info->capture.volume; + ucontrol->value.integer.value[0] = vol & 0xffff; + ucontrol->value.integer.value[1] = vol >> 16 & 0xffff; + break; + default: + dev_err(dev, "%s() invalid private_value=%ld\n", + __func__, kctrl->private_value); + return -EINVAL; + } + + return 0; +} + +static int siu_dai_put_volume(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) +{ + struct siu_port *port_info = snd_kcontrol_chip(kctrl); + struct device *dev = port_info->pcm->card->dev; + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + u32 new_vol; + u32 cur_vol; + + dev_dbg(dev, "%s\n", __func__); + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > SIU_MAX_VOLUME || + ucontrol->value.integer.value[1] < 0 || + ucontrol->value.integer.value[1] > SIU_MAX_VOLUME) + return -EINVAL; + + new_vol = ucontrol->value.integer.value[0] | + ucontrol->value.integer.value[1] << 16; + + /* See comment above - DSP firmware implementation */ + switch (kctrl->private_value) { + case VOLUME_PLAYBACK: + /* Playback is always on port 0 */ + cur_vol = port_info->playback.volume; + siu_write32(base + SIU_SBDVCA, new_vol); + port_info->playback.volume = new_vol; + break; + case VOLUME_CAPTURE: + /* Capture is always on port 1 */ + cur_vol = port_info->capture.volume; + siu_write32(base + SIU_SBDVCB, new_vol); + port_info->capture.volume = new_vol; + break; + default: + dev_err(dev, "%s() invalid private_value=%ld\n", + __func__, kctrl->private_value); + return -EINVAL; + } + + if (cur_vol != new_vol) + return 1; + + return 0; +} + +static struct snd_kcontrol_new playback_controls = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .info = siu_dai_info_volume, + .get = siu_dai_get_volume, + .put = siu_dai_put_volume, + .private_value = VOLUME_PLAYBACK, +}; + +static struct snd_kcontrol_new capture_controls = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Capture Volume", + .index = 0, + .info = siu_dai_info_volume, + .get = siu_dai_get_volume, + .put = siu_dai_put_volume, + .private_value = VOLUME_CAPTURE, +}; + +int siu_init_port(int port, struct siu_port **port_info, struct snd_card *card) +{ + struct device *dev = card->dev; + struct snd_kcontrol *kctrl; + int ret; + + *port_info = kzalloc(sizeof(**port_info), GFP_KERNEL); + if (!*port_info) + return -ENOMEM; + + dev_dbg(dev, "%s: port #%d@%p\n", __func__, port, *port_info); + + (*port_info)->playback.volume = DFLT_VOLUME_LEVEL; + (*port_info)->capture.volume = DFLT_VOLUME_LEVEL; + + /* + * Add mixer support. The SPB is used to change the volume. Both + * ports use the same SPB. Therefore, we only register one + * control instance since it will be used by both channels. + * In error case we continue without controls. + */ + kctrl = snd_ctl_new1(&playback_controls, *port_info); + ret = snd_ctl_add(card, kctrl); + if (ret < 0) + dev_err(dev, + "failed to add playback controls %p port=%d err=%d\n", + kctrl, port, ret); + + kctrl = snd_ctl_new1(&capture_controls, *port_info); + ret = snd_ctl_add(card, kctrl); + if (ret < 0) + dev_err(dev, + "failed to add capture controls %p port=%d err=%d\n", + kctrl, port, ret); + + return 0; +} + +void siu_free_port(struct siu_port *port_info) +{ + kfree(port_info); +} + +static int siu_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct snd_pcm_runtime *rt = substream->runtime; + struct siu_port *port_info = siu_port_info(substream); + int ret; + + dev_dbg(substream->pcm->card->dev, "%s: port=%d@%p\n", __func__, + info->port_id, port_info); + + snd_soc_set_runtime_hwparams(substream, &siu_dai_pcm_hw); + + ret = snd_pcm_hw_constraint_integer(rt, SNDRV_PCM_HW_PARAM_PERIODS); + if (unlikely(ret < 0)) + return ret; + + siu_dai_start(port_info); + + return 0; +} + +static void siu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_port *port_info = siu_port_info(substream); + + dev_dbg(substream->pcm->card->dev, "%s: port=%d@%p\n", __func__, + info->port_id, port_info); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_info->play_cap &= ~PLAYBACK_ENABLED; + else + port_info->play_cap &= ~CAPTURE_ENABLED; + + /* Stop the siu if the other stream is not using it */ + if (!port_info->play_cap) { + /* during stmread or stmwrite ? */ + BUG_ON(port_info->playback.rw_flg || port_info->capture.rw_flg); + siu_dai_spbstop(port_info); + siu_dai_stop(); + } +} + +/* PCM part of siu_dai_playback_prepare() / siu_dai_capture_prepare() */ +static int siu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct snd_pcm_runtime *rt = substream->runtime; + struct siu_port *port_info = siu_port_info(substream); + struct siu_stream *siu_stream; + int self, ret; + + dev_dbg(substream->pcm->card->dev, + "%s: port %d, active streams %lx, %d channels\n", + __func__, info->port_id, port_info->play_cap, rt->channels); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + self = PLAYBACK_ENABLED; + siu_stream = &port_info->playback; + } else { + self = CAPTURE_ENABLED; + siu_stream = &port_info->capture; + } + + /* Set up the siu if not already done */ + if (!port_info->play_cap) { + siu_stream->rw_flg = 0; /* stream-data transfer flag */ + + siu_dai_spbAselect(port_info); + siu_dai_spbBselect(port_info); + + siu_dai_open(siu_stream); + + siu_dai_pcmdatapack(siu_stream); + + ret = siu_dai_spbstart(port_info); + if (ret < 0) + goto fail; + } + + port_info->play_cap |= self; + +fail: + return ret; +} + +/* + * SIU can set bus format to I2S / PCM / SPDIF independently for playback and + * capture, however, the current API sets the bus format globally for a DAI. + */ +static int siu_dai_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + u32 ifctl; + + dev_dbg(dai->dev, "%s: fmt 0x%x on port %d\n", + __func__, fmt, info->port_id); + + if (info->port_id < 0) + return -ENODEV; + + /* Here select between I2S / PCM / SPDIF */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ifctl = siu_flags[info->port_id].playback.i2s | + siu_flags[info->port_id].capture.i2s; + break; + case SND_SOC_DAIFMT_LEFT_J: + ifctl = siu_flags[info->port_id].playback.pcm | + siu_flags[info->port_id].capture.pcm; + break; + /* SPDIF disabled - see comment at the top */ + default: + return -EINVAL; + } + + ifctl |= ~(siu_flags[info->port_id].playback.mask | + siu_flags[info->port_id].capture.mask) & + siu_read32(base + SIU_IFCTL); + siu_write32(base + SIU_IFCTL, ifctl); + + return 0; +} + +static int siu_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct clk *siu_clk, *parent_clk; + char *siu_name, *parent_name; + int ret; + + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + + dev_dbg(dai->dev, "%s: using clock %d\n", __func__, clk_id); + + switch (clk_id) { + case SIU_CLKA_PLL: + siu_name = "siua_clk"; + parent_name = "pll_clk"; + break; + case SIU_CLKA_EXT: + siu_name = "siua_clk"; + parent_name = "siumcka_clk"; + break; + case SIU_CLKB_PLL: + siu_name = "siub_clk"; + parent_name = "pll_clk"; + break; + case SIU_CLKB_EXT: + siu_name = "siub_clk"; + parent_name = "siumckb_clk"; + break; + default: + return -EINVAL; + } + + siu_clk = clk_get(siu_i2s_dai.dev, siu_name); + if (IS_ERR(siu_clk)) + return PTR_ERR(siu_clk); + + parent_clk = clk_get(siu_i2s_dai.dev, parent_name); + if (!IS_ERR(parent_clk)) { + ret = clk_set_parent(siu_clk, parent_clk); + if (!ret) + clk_set_rate(siu_clk, freq); + clk_put(parent_clk); + } + + clk_put(siu_clk); + + return 0; +} + +static struct snd_soc_dai_ops siu_dai_ops = { + .startup = siu_dai_startup, + .shutdown = siu_dai_shutdown, + .prepare = siu_dai_prepare, + .set_sysclk = siu_dai_set_sysclk, + .set_fmt = siu_dai_set_fmt, +}; + +struct snd_soc_dai siu_i2s_dai = { + .name = "sh-siu", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_8000_48000, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_8000_48000, + }, + .ops = &siu_dai_ops, +}; +EXPORT_SYMBOL_GPL(siu_i2s_dai); + +static int __devinit siu_probe(struct platform_device *pdev) +{ + const struct firmware *fw_entry; + struct resource *res, *region; + struct siu_info *info; + int ret; + + info = kmalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev); + if (ret) + goto ereqfw; + + /* + * Loaded firmware is "const" - read only, but we have to modify it in + * snd_siu_sh7343_spbAselect() and snd_siu_sh7343_spbBselect() + */ + memcpy(&info->fw, fw_entry->data, fw_entry->size); + + release_firmware(fw_entry); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + goto egetres; + } + + region = request_mem_region(res->start, resource_size(res), + pdev->name); + if (!region) { + dev_err(&pdev->dev, "SIU region already claimed\n"); + ret = -EBUSY; + goto ereqmemreg; + } + + ret = -ENOMEM; + info->pram = ioremap(res->start, PRAM_SIZE); + if (!info->pram) + goto emappram; + info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE); + if (!info->xram) + goto emapxram; + info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE); + if (!info->yram) + goto emapyram; + info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) - + REG_OFFSET); + if (!info->reg) + goto emapreg; + + siu_i2s_dai.dev = &pdev->dev; + siu_i2s_dai.private_data = info; + + ret = snd_soc_register_dais(&siu_i2s_dai, 1); + if (ret < 0) + goto edaiinit; + + ret = snd_soc_register_platform(&siu_platform); + if (ret < 0) + goto esocregp; + + pm_runtime_enable(&pdev->dev); + + return ret; + +esocregp: + snd_soc_unregister_dais(&siu_i2s_dai, 1); +edaiinit: + iounmap(info->reg); +emapreg: + iounmap(info->yram); +emapyram: + iounmap(info->xram); +emapxram: + iounmap(info->pram); +emappram: + release_mem_region(res->start, resource_size(res)); +ereqmemreg: +egetres: +ereqfw: + kfree(info); + + return ret; +} + +static int __devexit siu_remove(struct platform_device *pdev) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct resource *res; + + pm_runtime_disable(&pdev->dev); + + snd_soc_unregister_platform(&siu_platform); + snd_soc_unregister_dais(&siu_i2s_dai, 1); + + iounmap(info->reg); + iounmap(info->yram); + iounmap(info->xram); + iounmap(info->pram); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res) + release_mem_region(res->start, resource_size(res)); + kfree(info); + + return 0; +} + +static struct platform_driver siu_driver = { + .driver = { + .name = "sh_siu", + }, + .probe = siu_probe, + .remove = __devexit_p(siu_remove), +}; + +static int __init siu_init(void) +{ + return platform_driver_register(&siu_driver); +} + +static void __exit siu_exit(void) +{ + platform_driver_unregister(&siu_driver); +} + +module_init(siu_init) +module_exit(siu_exit) + +MODULE_AUTHOR("Carlos Munoz "); +MODULE_DESCRIPTION("ALSA SoC SH7722 SIU driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c new file mode 100644 index 0000000..c5efc30 --- /dev/null +++ b/sound/soc/sh/siu_pcm.c @@ -0,0 +1,616 @@ +/* + * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * Copyright (C) 2006 Carlos Munoz + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "siu.h" + +#define GET_MAX_PERIODS(buf_bytes, period_bytes) \ + ((buf_bytes) / (period_bytes)) +#define PERIOD_OFFSET(buf_addr, period_num, period_bytes) \ + ((buf_addr) + ((period_num) * (period_bytes))) + +#define RWF_STM_RD 0x01 /* Read in progress */ +#define RWF_STM_WT 0x02 /* Write in progress */ + +struct siu_port *siu_ports[SIU_PORT_NUM]; + +/* transfersize is number of u32 dma transfers per period */ +static int siu_pcm_stmwrite_stop(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_stream *siu_stream = &port_info->playback; + u32 stfifo; + + if (!siu_stream->rw_flg) + return -EPERM; + + /* output FIFO disable */ + stfifo = siu_read32(base + SIU_STFIFO); + siu_write32(base + SIU_STFIFO, stfifo & ~0x0c180c18); + pr_debug("%s: STFIFO %x -> %x\n", __func__, + stfifo, stfifo & ~0x0c180c18); + + /* during stmwrite clear */ + siu_stream->rw_flg = 0; + + return 0; +} + +static int siu_pcm_stmwrite_start(struct siu_port *port_info) +{ + struct siu_stream *siu_stream = &port_info->playback; + + if (siu_stream->rw_flg) + return -EPERM; + + /* Current period in buffer */ + port_info->playback.cur_period = 0; + + /* during stmwrite flag set */ + siu_stream->rw_flg = RWF_STM_WT; + + /* DMA transfer start */ + tasklet_schedule(&siu_stream->tasklet); + + return 0; +} + +static void siu_dma_tx_complete(void *arg) +{ + struct siu_stream *siu_stream = arg; + + if (!siu_stream->rw_flg) + return; + + /* Update completed period count */ + if (++siu_stream->cur_period >= + GET_MAX_PERIODS(siu_stream->buf_bytes, + siu_stream->period_bytes)) + siu_stream->cur_period = 0; + + pr_debug("%s: done period #%d (%u/%u bytes), cookie %d\n", + __func__, siu_stream->cur_period, + siu_stream->cur_period * siu_stream->period_bytes, + siu_stream->buf_bytes, siu_stream->cookie); + + tasklet_schedule(&siu_stream->tasklet); + + /* Notify alsa: a period is done */ + snd_pcm_period_elapsed(siu_stream->substream); +} + +static int siu_pcm_wr_set(struct siu_port *port_info, + dma_addr_t buff, u32 size) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_stream *siu_stream = &port_info->playback; + struct snd_pcm_substream *substream = siu_stream->substream; + struct device *dev = substream->pcm->card->dev; + struct dma_async_tx_descriptor *desc; + dma_cookie_t cookie; + struct scatterlist sg; + u32 stfifo; + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buff)), + size, offset_in_page(buff)); + sg_dma_address(&sg) = buff; + + desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan, + &sg, 1, DMA_TO_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dev, "Failed to allocate a dma descriptor\n"); + return -ENOMEM; + } + + desc->callback = siu_dma_tx_complete; + desc->callback_param = siu_stream; + cookie = desc->tx_submit(desc); + if (cookie < 0) { + dev_err(dev, "Failed to submit a dma transfer\n"); + return cookie; + } + + siu_stream->tx_desc = desc; + siu_stream->cookie = cookie; + + dma_async_issue_pending(siu_stream->chan); + + /* only output FIFO enable */ + stfifo = siu_read32(base + SIU_STFIFO); + siu_write32(base + SIU_STFIFO, stfifo | (port_info->stfifo & 0x0c180c18)); + dev_dbg(dev, "%s: STFIFO %x -> %x\n", __func__, + stfifo, stfifo | (port_info->stfifo & 0x0c180c18)); + + return 0; +} + +static int siu_pcm_rd_set(struct siu_port *port_info, + dma_addr_t buff, size_t size) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_stream *siu_stream = &port_info->capture; + struct snd_pcm_substream *substream = siu_stream->substream; + struct device *dev = substream->pcm->card->dev; + struct dma_async_tx_descriptor *desc; + dma_cookie_t cookie; + struct scatterlist sg; + u32 stfifo; + + dev_dbg(dev, "%s: %u@%llx\n", __func__, size, (unsigned long long)buff); + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buff)), + size, offset_in_page(buff)); + sg_dma_address(&sg) = buff; + + desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan, + &sg, 1, DMA_FROM_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dev, "Failed to allocate dma descriptor\n"); + return -ENOMEM; + } + + desc->callback = siu_dma_tx_complete; + desc->callback_param = siu_stream; + cookie = desc->tx_submit(desc); + if (cookie < 0) { + dev_err(dev, "Failed to submit dma descriptor\n"); + return cookie; + } + + siu_stream->tx_desc = desc; + siu_stream->cookie = cookie; + + dma_async_issue_pending(siu_stream->chan); + + /* only input FIFO enable */ + stfifo = siu_read32(base + SIU_STFIFO); + siu_write32(base + SIU_STFIFO, siu_read32(base + SIU_STFIFO) | + (port_info->stfifo & 0x13071307)); + dev_dbg(dev, "%s: STFIFO %x -> %x\n", __func__, + stfifo, stfifo | (port_info->stfifo & 0x13071307)); + + return 0; +} + +static void siu_io_tasklet(unsigned long data) +{ + struct siu_stream *siu_stream = (struct siu_stream *)data; + struct snd_pcm_substream *substream = siu_stream->substream; + struct device *dev = substream->pcm->card->dev; + struct snd_pcm_runtime *rt = substream->runtime; + struct siu_port *port_info = siu_port_info(substream); + + dev_dbg(dev, "%s: flags %x\n", __func__, siu_stream->rw_flg); + + if (!siu_stream->rw_flg) { + dev_dbg(dev, "%s: stream inactive\n", __func__); + return; + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + dma_addr_t buff; + size_t count; + u8 *virt; + + buff = (dma_addr_t)PERIOD_OFFSET(rt->dma_addr, + siu_stream->cur_period, + siu_stream->period_bytes); + virt = PERIOD_OFFSET(rt->dma_area, + siu_stream->cur_period, + siu_stream->period_bytes); + count = siu_stream->period_bytes; + + /* DMA transfer start */ + siu_pcm_rd_set(port_info, buff, count); + } else { + siu_pcm_wr_set(port_info, + (dma_addr_t)PERIOD_OFFSET(rt->dma_addr, + siu_stream->cur_period, + siu_stream->period_bytes), + siu_stream->period_bytes); + } +} + +/* Capture */ +static int siu_pcm_stmread_start(struct siu_port *port_info) +{ + struct siu_stream *siu_stream = &port_info->capture; + + if (siu_stream->xfer_cnt > 0x1000000) + return -EINVAL; + if (siu_stream->rw_flg) + return -EPERM; + + /* Current period in buffer */ + siu_stream->cur_period = 0; + + /* during stmread flag set */ + siu_stream->rw_flg = RWF_STM_RD; + + tasklet_schedule(&siu_stream->tasklet); + + return 0; +} + +static int siu_pcm_stmread_stop(struct siu_port *port_info) +{ + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_stream *siu_stream = &port_info->capture; + struct device *dev = siu_stream->substream->pcm->card->dev; + u32 stfifo; + + if (!siu_stream->rw_flg) + return -EPERM; + + /* input FIFO disable */ + stfifo = siu_read32(base + SIU_STFIFO); + siu_write32(base + SIU_STFIFO, stfifo & ~0x13071307); + dev_dbg(dev, "%s: STFIFO %x -> %x\n", __func__, + stfifo, stfifo & ~0x13071307); + + /* during stmread flag clear */ + siu_stream->rw_flg = 0; + + return 0; +} + +static int siu_pcm_hw_params(struct snd_pcm_substream *ss, + struct snd_pcm_hw_params *hw_params) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct device *dev = ss->pcm->card->dev; + int ret; + + dev_dbg(dev, "%s: port=%d\n", __func__, info->port_id); + + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(hw_params)); + if (ret < 0) + dev_err(dev, "snd_pcm_lib_malloc_pages() failed\n"); + + return ret; +} + +static int siu_pcm_hw_free(struct snd_pcm_substream *ss) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_port *port_info = siu_port_info(ss); + struct device *dev = ss->pcm->card->dev; + struct siu_stream *siu_stream; + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_stream = &port_info->playback; + else + siu_stream = &port_info->capture; + + dev_dbg(dev, "%s: port=%d\n", __func__, info->port_id); + + return snd_pcm_lib_free_pages(ss); +} + +static bool filter(struct dma_chan *chan, void *slave) +{ + struct sh_dmae_slave *param = slave; + + pr_debug("%s: slave ID %d\n", __func__, param->slave_id); + + if (unlikely(param->dma_dev != chan->device->dev)) + return false; + + chan->private = param; + return true; +} + +static int siu_pcm_open(struct snd_pcm_substream *ss) +{ + /* Playback / Capture */ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_port *port_info = siu_port_info(ss); + struct siu_stream *siu_stream; + u32 port = info->port_id; + struct siu_platform *pdata = siu_i2s_dai.dev->platform_data; + struct device *dev = ss->pcm->card->dev; + dma_cap_mask_t mask; + struct sh_dmae_slave *param; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + dev_dbg(dev, "%s, port=%d@%p\n", __func__, port, port_info); + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) { + siu_stream = &port_info->playback; + param = &siu_stream->param; + param->slave_id = port ? SHDMA_SLAVE_SIUB_TX : + SHDMA_SLAVE_SIUA_TX; + } else { + siu_stream = &port_info->capture; + param = &siu_stream->param; + param->slave_id = port ? SHDMA_SLAVE_SIUB_RX : + SHDMA_SLAVE_SIUA_RX; + } + + param->dma_dev = pdata->dma_dev; + /* Get DMA channel */ + siu_stream->chan = dma_request_channel(mask, filter, param); + if (!siu_stream->chan) { + dev_err(dev, "DMA channel allocation failed!\n"); + return -EBUSY; + } + + siu_stream->substream = ss; + + return 0; +} + +static int siu_pcm_close(struct snd_pcm_substream *ss) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct device *dev = ss->pcm->card->dev; + struct siu_port *port_info = siu_port_info(ss); + struct siu_stream *siu_stream; + + dev_dbg(dev, "%s: port=%d\n", __func__, info->port_id); + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_stream = &port_info->playback; + else + siu_stream = &port_info->capture; + + dma_release_channel(siu_stream->chan); + siu_stream->chan = NULL; + + siu_stream->substream = NULL; + + return 0; +} + +static int siu_pcm_prepare(struct snd_pcm_substream *ss) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct siu_port *port_info = siu_port_info(ss); + struct device *dev = ss->pcm->card->dev; + struct snd_pcm_runtime *rt = ss->runtime; + struct siu_stream *siu_stream; + snd_pcm_sframes_t xfer_cnt; + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_stream = &port_info->playback; + else + siu_stream = &port_info->capture; + + rt = siu_stream->substream->runtime; + + siu_stream->buf_bytes = snd_pcm_lib_buffer_bytes(ss); + siu_stream->period_bytes = snd_pcm_lib_period_bytes(ss); + + dev_dbg(dev, "%s: port=%d, %d channels, period=%u bytes\n", __func__, + info->port_id, rt->channels, siu_stream->period_bytes); + + /* We only support buffers that are multiples of the period */ + if (siu_stream->buf_bytes % siu_stream->period_bytes) { + dev_err(dev, "%s() - buffer=%d not multiple of period=%d\n", + __func__, siu_stream->buf_bytes, + siu_stream->period_bytes); + return -EINVAL; + } + + xfer_cnt = bytes_to_frames(rt, siu_stream->period_bytes); + if (!xfer_cnt || xfer_cnt > 0x1000000) + return -EINVAL; + + siu_stream->format = rt->format; + siu_stream->xfer_cnt = xfer_cnt; + + dev_dbg(dev, "port=%d buf=%lx buf_bytes=%d period_bytes=%d " + "format=%d channels=%d xfer_cnt=%d\n", info->port_id, + (unsigned long)rt->dma_addr, siu_stream->buf_bytes, + siu_stream->period_bytes, + siu_stream->format, rt->channels, (int)xfer_cnt); + + return 0; +} + +static int siu_pcm_trigger(struct snd_pcm_substream *ss, int cmd) +{ + struct siu_info *info = siu_i2s_dai.private_data; + struct device *dev = ss->pcm->card->dev; + struct siu_port *port_info = siu_port_info(ss); + int ret; + + dev_dbg(dev, "%s: port=%d@%p, cmd=%d\n", __func__, + info->port_id, port_info, cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = siu_pcm_stmwrite_start(port_info); + else + ret = siu_pcm_stmread_start(port_info); + + if (ret < 0) + dev_warn(dev, "%s: start failed on port=%d\n", + __func__, info->port_id); + + break; + case SNDRV_PCM_TRIGGER_STOP: + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_pcm_stmwrite_stop(port_info); + else + siu_pcm_stmread_stop(port_info); + ret = 0; + + break; + default: + dev_err(dev, "%s() unsupported cmd=%d\n", __func__, cmd); + ret = -EINVAL; + } + + return ret; +} + +/* + * So far only resolution of one period is supported, subject to extending the + * dmangine API + */ +static snd_pcm_uframes_t siu_pcm_pointer_dma(struct snd_pcm_substream *ss) +{ + struct device *dev = ss->pcm->card->dev; + struct siu_info *info = siu_i2s_dai.private_data; + u32 __iomem *base = info->reg; + struct siu_port *port_info = siu_port_info(ss); + struct snd_pcm_runtime *rt = ss->runtime; + size_t ptr; + struct siu_stream *siu_stream; + + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + siu_stream = &port_info->playback; + else + siu_stream = &port_info->capture; + + /* + * ptr is the offset into the buffer where the dma is currently at. We + * check if the dma buffer has just wrapped. + */ + ptr = PERIOD_OFFSET(rt->dma_addr, + siu_stream->cur_period, + siu_stream->period_bytes) - rt->dma_addr; + + dev_dbg(dev, + "%s: port=%d, events %x, FSTS %x, xferred %u/%u, cookie %d\n", + __func__, info->port_id, siu_read32(base + SIU_EVNTC), + siu_read32(base + SIU_SBFSTS), ptr, siu_stream->buf_bytes, + siu_stream->cookie); + + if (ptr >= siu_stream->buf_bytes) + ptr = 0; + + return bytes_to_frames(ss->runtime, ptr); +} + +static int siu_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + /* card->dev == socdev->dev, see snd_soc_new_pcms() */ + struct siu_info *info = siu_i2s_dai.private_data; + struct platform_device *pdev = to_platform_device(card->dev); + int ret; + int i; + + /* pdev->id selects between SIUA and SIUB */ + if (pdev->id < 0 || pdev->id >= SIU_PORT_NUM) + return -EINVAL; + + info->port_id = pdev->id; + + /* + * While the siu has 2 ports, only one port can be on at a time (only 1 + * SPB). So far all the boards using the siu had only one of the ports + * wired to a codec. To simplify things, we only register one port with + * alsa. In case both ports are needed, it should be changed here + */ + for (i = pdev->id; i < pdev->id + 1; i++) { + struct siu_port **port_info = &siu_ports[i]; + + ret = siu_init_port(i, port_info, card); + if (ret < 0) + return ret; + + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, NULL, + SIU_BUFFER_BYTES_MAX, SIU_BUFFER_BYTES_MAX); + if (ret < 0) { + dev_err(card->dev, + "snd_pcm_lib_preallocate_pages_for_all() err=%d", + ret); + goto fail; + } + + (*port_info)->pcm = pcm; + + /* IO tasklets */ + tasklet_init(&(*port_info)->playback.tasklet, siu_io_tasklet, + (unsigned long)&(*port_info)->playback); + tasklet_init(&(*port_info)->capture.tasklet, siu_io_tasklet, + (unsigned long)&(*port_info)->capture); + } + + dev_info(card->dev, "SuperH SIU driver initialized.\n"); + return 0; + +fail: + siu_free_port(siu_ports[pdev->id]); + dev_err(card->dev, "SIU: failed to initialize.\n"); + return ret; +} + +static void siu_pcm_free(struct snd_pcm *pcm) +{ + struct platform_device *pdev = to_platform_device(pcm->card->dev); + struct siu_port *port_info = siu_ports[pdev->id]; + + tasklet_kill(&port_info->capture.tasklet); + tasklet_kill(&port_info->playback.tasklet); + + siu_free_port(port_info); + snd_pcm_lib_preallocate_free_for_all(pcm); + + dev_dbg(pcm->card->dev, "%s\n", __func__); +} + +static struct snd_pcm_ops siu_pcm_ops = { + .open = siu_pcm_open, + .close = siu_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = siu_pcm_hw_params, + .hw_free = siu_pcm_hw_free, + .prepare = siu_pcm_prepare, + .trigger = siu_pcm_trigger, + .pointer = siu_pcm_pointer_dma, +}; + +struct snd_soc_platform siu_platform = { + .name = "siu-audio", + .pcm_ops = &siu_pcm_ops, + .pcm_new = siu_pcm_new, + .pcm_free = siu_pcm_free, +}; +EXPORT_SYMBOL_GPL(siu_platform); -- cgit v0.10.2 From 84549d239ab9bb2e3a85c6efcf0e6478a38b4260 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Mon, 25 Jan 2010 16:42:25 +0800 Subject: ASoC: ad1836: reset and restore clock control mode in suspend/resume entry tests show frequent suspend/resume(frequent poweroff/on ad1836 internal components) maybe make ad1836 clock mode wrong sometimes after wakeup. This patch reset/restore ad1836 clock mode while executing PM, then ad1836 can always resume to right clock status. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 2c18e3d..83add2f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,6 +223,36 @@ static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, return reg_cache[reg]; } +#ifdef CONFIG_PM +static int ad1836_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* reset clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} + +static int ad1836_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* restore clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 |= AD1836_ADC_AUX; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} +#else +#define ad1836_soc_suspend NULL +#define ad1836_soc_resume NULL +#endif + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -404,6 +434,8 @@ static int ad1836_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, + .suspend = ad1836_soc_suspend, + .resume = ad1836_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 7660ee6..e9d90d3 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -54,6 +54,7 @@ #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) +#define AD1836_ADC_AUX (0x6 << 6) #define AD1836_ADC_CTRL3 14 -- cgit v0.10.2 From cf944ee55cc318bdb1d4b2f3f5cce3257f7c07b3 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Tue, 26 Jan 2010 09:06:14 +0100 Subject: ALSA: cs46xx: Fix cpu idling with resume Make sure that capture DMA doesn't stay enabled after system resume as that potentially prevents the processor from entering deep sleep states. Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index e6b4a87..56fcf00 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -3644,6 +3644,7 @@ int snd_cs46xx_resume(struct pci_dev *pci) #ifdef CONFIG_SND_CS46XX_NEW_DSP int i; #endif + unsigned int tmp; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -3685,6 +3686,15 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* + * Stop capture DMA. + */ + tmp = snd_cs46xx_peek(chip, BA1_CCTL); + chip->capt.ctl = tmp & 0x0000ffff; + snd_cs46xx_poke(chip, BA1_CCTL, tmp & 0xffff0000); + + mdelay(5); + /* reset playback/capture */ snd_cs46xx_set_play_sample_rate(chip, 8000); snd_cs46xx_set_capture_sample_rate(chip, 8000); -- cgit v0.10.2 From ccc5df058da70d1c26c72cd1c24072a89998d735 Mon Sep 17 00:00:00 2001 From: Wei Ni Date: Tue, 26 Jan 2010 15:59:33 +0800 Subject: ALSA: hda - Add support for more the 8 streams In azx_stream_start() and azx_stream_stop(), it use azx_readb/azx_writeb to read/write SIE, it just enable/disable 8 streams. But according to the HDA spec, it support 30 streams, and the new HDA controller will support more then 8 streams. So we should use azx_readl/azx_writel to read/write SIE. Signed-off-by: Wei Ni Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6d331c4..6eeefda 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -954,8 +954,8 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev) azx_dev->insufficient = 1; /* enable SIE */ - azx_writeb(chip, INTCTL, - azx_readb(chip, INTCTL) | (1 << azx_dev->index)); + azx_writel(chip, INTCTL, + azx_readl(chip, INTCTL) | (1 << azx_dev->index)); /* set DMA start and interrupt mask */ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_DMA_START | SD_INT_MASK); @@ -974,8 +974,8 @@ static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) { azx_stream_clear(chip, azx_dev); /* disable SIE */ - azx_writeb(chip, INTCTL, - azx_readb(chip, INTCTL) & ~(1 << azx_dev->index)); + azx_writel(chip, INTCTL, + azx_readl(chip, INTCTL) & ~(1 << azx_dev->index)); } -- cgit v0.10.2 From e473b847424bd215b686cbc1e781e82c904ee967 Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Wed, 20 Jan 2010 17:06:33 +0530 Subject: ASoC: DaVinci: Fix stream restart error Sometimes after a suspend-resume cycle, the ALSA application restarts the stream when resume fails and McASP fails to work as the clock is not enabled. This patch corrects this bug. Testes on TI DA850/OMAP-L138 EVM. Signed-off-by: Chaithrika U S Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index a613bbb..ab6518d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -768,13 +768,12 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!dev->clk_active) { clk_enable(dev->clk); dev->clk_active = 1; } - /* Fall through */ - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; -- cgit v0.10.2 From e7636925789b042ff9d98c51d48392e8c5549480 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 26 Jan 2010 17:08:24 +0100 Subject: ALSA: pcm_lib - return back hw_ptr_interrupt Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr update functions" commit: "It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data." Return back the hw_ptr_interrupt variable. The last interrupt pointer is always computed from the latest hw_ptr instead of tracking it separately (in this case all hw_ptr checks and modifications might influence also hw_ptr_interrupt and it is difficult to keep it consistent). Signed-off-by: Jaroslav Kysela diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3bc9bca..13bc83c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -271,6 +271,7 @@ struct snd_pcm_runtime { int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ + snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 255ad91..82d4e33 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -635,8 +635,7 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) static inline snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) { - snd_pcm_uframes_t ptr = runtime->status->hw_ptr; - return ptr - (ptr % runtime->period_size); + return runtime->hw_ptr_interrupt; } /* define extended formats in the recent OSS versions (if any) */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e2a817e..aa54195 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -325,8 +325,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ - delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) - + runtime->period_size; + delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) @@ -437,6 +436,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + if (in_interrupt) { + runtime->hw_ptr_interrupt = new_hw_ptr - + (new_hw_ptr % runtime->period_size); + } runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 56ec35e..7a002db 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1252,6 +1252,8 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) if (err < 0) return err; runtime->hw_ptr_base = 0; + runtime->hw_ptr_interrupt = runtime->status->hw_ptr - + runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; return 0; -- cgit v0.10.2 From 63b62ab0d52c736b3274b294df962e0a4b7aae79 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Wed, 27 Jan 2010 11:46:17 +0800 Subject: ASoC: ad1836: use soc-cache framework for codec registers access Signed-off-by: Barry Song Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 83add2f..3c80137 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -171,58 +171,6 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, return 0; } - -/* - * interface to read/write ad1836 register - */ -#define AD1836_SPI_REG_SHFT 12 -#define AD1836_SPI_READ (1 << 11) -#define AD1836_SPI_VAL_MSK 0x3FF - -/* - * write to the ad1836 register space - */ - -static int ad1836_write_reg(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 *reg_cache = codec->reg_cache; - int ret = 0; - - if (value != reg_cache[reg]) { - unsigned short buf; - struct spi_transfer t = { - .tx_buf = &buf, - .len = 2, - }; - struct spi_message m; - - buf = (reg << AD1836_SPI_REG_SHFT) | - (value & AD1836_SPI_VAL_MSK); - spi_message_init(&m); - spi_message_add_tail(&t, &m); - ret = spi_sync(codec->control_data, &m); - if (ret == 0) - reg_cache[reg] = value; - } - - return ret; -} - -/* - * read from the ad1836 register space cache - */ -static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *reg_cache = codec->reg_cache; - - if (reg >= codec->reg_cache_size) - return -EINVAL; - - return reg_cache[reg]; -} - #ifdef CONFIG_PM static int ad1836_soc_suspend(struct platform_device *pdev, pm_message_t state) @@ -231,10 +179,10 @@ static int ad1836_soc_suspend(struct platform_device *pdev, struct snd_soc_codec *codec = socdev->card->codec; /* reset clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); } static int ad1836_soc_resume(struct platform_device *pdev) @@ -243,10 +191,10 @@ static int ad1836_soc_resume(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->card->codec; /* restore clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); adc_ctrl2 |= AD1836_ADC_AUX; - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); } #else #define ad1836_soc_suspend NULL @@ -336,32 +284,38 @@ static int ad1836_register(struct ad1836_priv *ad1836) codec->owner = THIS_MODULE; codec->dai = &ad1836_dai; codec->num_dai = 1; - codec->write = ad1836_write_reg; - codec->read = ad1836_read_reg_cache; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ad1836_dai.dev = codec->dev; ad1836_codec = codec; + ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", + ret); + kfree(ad1836); + return ret; + } + /* default setting for ad1836 */ /* de-emphasis: 48kHz, power-on dac */ - codec->write(codec, AD1836_DAC_CTRL1, 0x300); + snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300); /* unmute dac channels */ - codec->write(codec, AD1836_DAC_CTRL2, 0x0); + snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0); /* high-pass filter enable, power-on adc */ - codec->write(codec, AD1836_ADC_CTRL1, 0x100); + snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); /* unmute adc channles, adc aux mode */ - codec->write(codec, AD1836_ADC_CTRL2, 0x180); + snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); /* left/right diff:PGA/MUX */ - codec->write(codec, AD1836_ADC_CTRL3, 0x3A); + snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); /* volume */ - codec->write(codec, AD1836_DAC_L1_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R1_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_L2_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R2_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_L3_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R3_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF); ret = snd_soc_register_codec(codec); if (ret != 0) { diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 02c2357..cde7b63 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -15,6 +15,68 @@ #include #include +static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + +static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[2]; + int ret; + + BUG_ON(codec->volatile_register); + + data[0] = (reg << 4) | ((value >> 8) & 0x000f); + data[1] = value & 0x00ff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +#if defined(CONFIG_SPI_MASTER) +static int snd_soc_4_12_spi_write(void *control_data, const char *data, + int len) +{ + struct spi_device *spi = control_data; + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[1]; + msg[1] = data[0]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#else +#define snd_soc_4_12_spi_write NULL +#endif + static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -180,6 +242,11 @@ static struct { unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { + .addr_bits = 4, .data_bits = 12, + .write = snd_soc_4_12_write, .read = snd_soc_4_12_read, + .spi_write = snd_soc_4_12_spi_write, + }, + { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, .spi_write = snd_soc_7_9_spi_write, -- cgit v0.10.2 From 994dc4245d3f50329da4ead453a5dfcfbc716a0d Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Wed, 27 Jan 2010 11:46:18 +0800 Subject: ASoC: ad1938: use soc-cache framework for codec registers access Signed-off-by: Barry Song Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 47d9ac0..c233810 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -46,6 +46,11 @@ struct ad1938_priv { u8 reg_cache[AD1938_NUM_REGS]; }; +/* ad1938 register cache & default register settings */ +static const u8 ad1938_reg[AD1938_NUM_REGS] = { + 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0, +}; + static struct snd_soc_codec *ad1938_codec; struct snd_soc_codec_device soc_codec_dev_ad1938; static int ad1938_register(struct ad1938_priv *ad1938); @@ -129,10 +134,10 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; int reg; - reg = codec->read(codec, AD1938_DAC_CTRL2); + reg = snd_soc_read(codec, AD1938_DAC_CTRL2); reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & (~AD1938_DAC_MASTER_MUTE); - codec->write(codec, AD1938_DAC_CTRL2, reg); + snd_soc_write(codec, AD1938_DAC_CTRL2, reg); return 0; } @@ -141,8 +146,8 @@ static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int width) { struct snd_soc_codec *codec = dai->codec; - int dac_reg = codec->read(codec, AD1938_DAC_CTRL1); - int adc_reg = codec->read(codec, AD1938_ADC_CTRL2); + int dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1); + int adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2); dac_reg &= ~AD1938_DAC_CHAN_MASK; adc_reg &= ~AD1938_ADC_CHAN_MASK; @@ -168,8 +173,8 @@ static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return -EINVAL; } - codec->write(codec, AD1938_DAC_CTRL1, dac_reg); - codec->write(codec, AD1938_ADC_CTRL2, adc_reg); + snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg); + snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg); return 0; } @@ -180,8 +185,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; int adc_reg, dac_reg; - adc_reg = codec->read(codec, AD1938_ADC_CTRL2); - dac_reg = codec->read(codec, AD1938_DAC_CTRL1); + adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2); + dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1); /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) @@ -258,8 +263,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - codec->write(codec, AD1938_ADC_CTRL2, adc_reg); - codec->write(codec, AD1938_DAC_CTRL1, dac_reg); + snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg); + snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg); return 0; } @@ -288,116 +293,13 @@ static int ad1938_hw_params(struct snd_pcm_substream *substream, break; } - reg = codec->read(codec, AD1938_DAC_CTRL2); + reg = snd_soc_read(codec, AD1938_DAC_CTRL2); reg = (reg & (~AD1938_DAC_WORD_LEN_MASK)) | word_len; - codec->write(codec, AD1938_DAC_CTRL2, reg); + snd_soc_write(codec, AD1938_DAC_CTRL2, reg); - reg = codec->read(codec, AD1938_ADC_CTRL1); + reg = snd_soc_read(codec, AD1938_ADC_CTRL1); reg = (reg & (~AD1938_ADC_WORD_LEN_MASK)) | word_len; - codec->write(codec, AD1938_ADC_CTRL1, reg); - - return 0; -} - -/* - * interface to read/write ad1938 register - */ - -#define AD1938_SPI_ADDR 0x4 -#define AD1938_SPI_READ 0x1 -#define AD1938_SPI_BUFLEN 3 - -/* - * write to the ad1938 register space - */ - -static int ad1938_write_reg(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 *reg_cache = codec->reg_cache; - int ret = 0; - - if (value != reg_cache[reg]) { - uint8_t buf[AD1938_SPI_BUFLEN]; - struct spi_transfer t = { - .tx_buf = buf, - .len = AD1938_SPI_BUFLEN, - }; - struct spi_message m; - - buf[0] = AD1938_SPI_ADDR << 1; - buf[1] = reg; - buf[2] = value; - spi_message_init(&m); - spi_message_add_tail(&t, &m); - ret = spi_sync(codec->control_data, &m); - if (ret == 0) - reg_cache[reg] = value; - } - - return ret; -} - -/* - * read from the ad1938 register space cache - */ - -static unsigned int ad1938_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u8 *reg_cache = codec->reg_cache; - - if (reg >= codec->reg_cache_size) - return -EINVAL; - - return reg_cache[reg]; -} - -/* - * read from the ad1938 register space - */ - -static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, - unsigned int reg) -{ - char w_buf[AD1938_SPI_BUFLEN]; - char r_buf[AD1938_SPI_BUFLEN]; - int ret; - - struct spi_transfer t = { - .tx_buf = w_buf, - .rx_buf = r_buf, - .len = AD1938_SPI_BUFLEN, - }; - struct spi_message m; - - w_buf[0] = (AD1938_SPI_ADDR << 1) | AD1938_SPI_READ; - w_buf[1] = reg; - w_buf[2] = 0; - - spi_message_init(&m); - spi_message_add_tail(&t, &m); - ret = spi_sync(codec->control_data, &m); - if (ret == 0) - return r_buf[2]; - else - return -EIO; -} - -static int ad1938_fill_cache(struct snd_soc_codec *codec) -{ - int i; - u8 *reg_cache = codec->reg_cache; - struct spi_device *spi = codec->control_data; - - for (i = 0; i < codec->reg_cache_size; i++) { - int ret = ad1938_read_reg(codec, i); - if (ret == -EIO) { - dev_err(&spi->dev, "AD1938 SPI read failure\n"); - return ret; - } - reg_cache[i] = ret; - } + snd_soc_write(codec, AD1938_ADC_CTRL1, reg); return 0; } @@ -487,31 +389,37 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->owner = THIS_MODULE; codec->dai = &ad1938_dai; codec->num_dai = 1; - codec->write = ad1938_write_reg; - codec->read = ad1938_read_reg_cache; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ad1938_dai.dev = codec->dev; ad1938_codec = codec; + memcpy(codec->reg_cache, ad1938_reg, AD1938_NUM_REGS); + + ret = snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_SPI); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", + ret); + kfree(ad1938); + return ret; + } + /* default setting for ad1938 */ /* unmute dac channels */ - codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); + snd_soc_write(codec, AD1938_DAC_CHNL_MUTE, 0x0); /* de-emphasis: 48kHz, powedown dac */ - codec->write(codec, AD1938_DAC_CTRL2, 0x1A); + snd_soc_write(codec, AD1938_DAC_CTRL2, 0x1A); /* powerdown dac, dac in tdm mode */ - codec->write(codec, AD1938_DAC_CTRL0, 0x41); + snd_soc_write(codec, AD1938_DAC_CTRL0, 0x41); /* high-pass filter enable */ - codec->write(codec, AD1938_ADC_CTRL0, 0x3); + snd_soc_write(codec, AD1938_ADC_CTRL0, 0x3); /* sata delay=1, adc aux mode */ - codec->write(codec, AD1938_ADC_CTRL1, 0x43); + snd_soc_write(codec, AD1938_ADC_CTRL1, 0x43); /* pll input: mclki/xi */ - codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); - codec->write(codec, AD1938_PLL_CLK_CTRL1, 0x04); - - ad1938_fill_cache(codec); + snd_soc_write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); + snd_soc_write(codec, AD1938_PLL_CLK_CTRL1, 0x04); ret = snd_soc_register_codec(codec); if (ret != 0) { diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index cde7b63..097e335 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -233,6 +233,108 @@ static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, #define snd_soc_8_16_read_i2c NULL #endif +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + struct i2c_msg xfer[2]; + u16 reg = r; + u8 data; + int ret; + struct i2c_client *client = codec->control_data; + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 2; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 1; + xfer[1].buf = &data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return data; +} +#else +#define snd_soc_16_8_read_i2c NULL +#endif + +static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + reg &= 0xff; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + +static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[3]; + int ret; + + BUG_ON(codec->volatile_register); + + data[0] = (reg >> 8) & 0xff; + data[1] = reg & 0xff; + data[2] = value; + + reg &= 0xff; + if (reg < codec->reg_cache_size) + cache[reg] = value; + ret = codec->hw_write(codec->control_data, data, 3); + if (ret == 3) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +#if defined(CONFIG_SPI_MASTER) +static int snd_soc_16_8_spi_write(void *control_data, const char *data, + int len) +{ + struct spi_device *spi = control_data; + struct spi_transfer t; + struct spi_message m; + u8 msg[3]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + msg[2] = data[2]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#else +#define snd_soc_16_8_spi_write NULL +#endif + + static struct { int addr_bits; int data_bits; @@ -260,6 +362,12 @@ static struct { .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, .i2c_read = snd_soc_8_16_read_i2c, }, + { + .addr_bits = 16, .data_bits = 8, + .write = snd_soc_16_8_write, .read = snd_soc_16_8_read, + .i2c_read = snd_soc_16_8_read_i2c, + .spi_write = snd_soc_16_8_spi_write, + }, }; /** -- cgit v0.10.2 From 7910b4a1db63fefc3d291853d33c34c5b6352e8e Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 27 Jan 2010 18:10:13 +0100 Subject: ALSA: pcm_native - fix runtime->boundary calculation The code in pcm_lib updating runtime->hw_ptr_interrupt expects that runtime->boundary is divisible with runtime->period_size. Thanks are going to Clemens Ladisch for the notice. Fix the runtime->boundary calculation using buffer_size * period_size as base and find a least common multiple for 32bit platforms when the expression might overflow. Signed-off-by: Jaroslav Kysela diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7a002db..9cbaf90 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include @@ -366,6 +367,38 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) return usecs; } +static int calc_boundary(struct snd_pcm_runtime *runtime) +{ + u_int64_t boundary; + + boundary = (u_int64_t)runtime->buffer_size * + (u_int64_t)runtime->period_size; +#if BITS_PER_LONG < 64 + /* try to find lowest common multiple for buffer and period */ + if (boundary > LONG_MAX - runtime->buffer_size) { + u_int32_t remainder = -1; + u_int32_t divident = runtime->buffer_size; + u_int32_t divisor = runtime->period_size; + while (remainder) { + remainder = divident % divisor; + if (remainder) { + divident = divisor; + divisor = remainder; + } + } + boundary = div_u64(boundary, divisor); + if (boundary > LONG_MAX - runtime->buffer_size) + return -ERANGE; + } +#endif + if (boundary == 0) + return -ERANGE; + runtime->boundary = boundary; + while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) + runtime->boundary *= 2; + return 0; +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -441,9 +474,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->stop_threshold = runtime->buffer_size; runtime->silence_threshold = 0; runtime->silence_size = 0; - runtime->boundary = runtime->buffer_size; - while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) - runtime->boundary *= 2; + err = calc_boundary(runtime); + if (err < 0) + goto _error; snd_pcm_timer_resolution_change(substream); runtime->status->state = SNDRV_PCM_STATE_SETUP; -- cgit v0.10.2 From fc93ea2f9315eda2ec8645c2f8bcc30f75a6b88e Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 14:59:08 +0900 Subject: ASoC: AC97: S3C: Add controller driver Add the AC97 controller driver for Samsung SoCs that have one. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b489f1a..ad3690e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -32,7 +32,11 @@ config SND_S3C2443_SOC_AC97 select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_BUS - + +config SND_S3C_SOC_AC97 + tristate + select SND_SOC_AC97_BUS + config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b744657..b7411bd 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -4,12 +4,14 @@ snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o +snd-soc-s3c-ac97-objs := s3c-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o +obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o @@ -37,4 +39,3 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o - diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c new file mode 100644 index 0000000..ee8ed9d --- /dev/null +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -0,0 +1,518 @@ +/* sound/soc/s3c24xx/s3c-ac97.c + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.c + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include + +#include + +#include +#include +#include + +#include "s3c-dma.h" +#include "s3c-ac97.h" + +#define AC_CMD_ADDR(x) (x << 16) +#define AC_CMD_DATA(x) (x & 0xffff) + +struct s3c_ac97_info { + unsigned state; + struct clk *ac97_clk; + void __iomem *regs; + struct mutex lock; + struct completion done; +}; +static struct s3c_ac97_info s3c_ac97; + +static struct s3c2410_dma_client s3c_dma_client_out = { + .name = "AC97 PCMOut" +}; + +static struct s3c2410_dma_client s3c_dma_client_in = { + .name = "AC97 PCMIn" +}; + +static struct s3c2410_dma_client s3c_dma_client_micin = { + .name = "AC97 MicIn" +}; + +static struct s3c_dma_params s3c_ac97_pcm_out = { + .client = &s3c_dma_client_out, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_pcm_in = { + .client = &s3c_dma_client_in, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_mic_in = { + .client = &s3c_dma_client_micin, + .dma_size = 4, +}; + +static void s3c_ac97_activate(struct snd_ac97 *ac97) +{ + u32 ac_glbctrl, stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + INIT_COMPLETION(s3c_ac97.done); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to activate!"); +} + +static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + u32 ac_glbctrl, ac_codec_cmd; + u32 stat, addr, data; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to read!"); + + stat = readl(s3c_ac97.regs + S3C_AC97_STAT); + addr = (stat >> 16) & 0x7f; + data = (stat & 0xffff); + + if (addr != reg) + printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); + + mutex_unlock(&s3c_ac97.lock); + + return (unsigned short)data; +} + +static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + u32 ac_glbctrl, ac_codec_cmd; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to write!"); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + mutex_unlock(&s3c_ac97.lock); +} + +static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) +{ + writel(S3C_AC97_GLBCTRL_COLDRESET, + s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); +} + +static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) +{ + u32 stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + s3c_ac97_activate(ac97); +} + +static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) +{ + u32 ac_glbctrl, ac_glbstat; + + ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); + + if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + complete(&s3c_ac97.done); + } + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= (1<<30); /* Clear interrupt */ + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + return IRQ_HANDLED; +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = s3c_ac97_read, + .write = s3c_ac97_write, + .warm_reset = s3c_ac97_warm_reset, + .reset = s3c_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_dai->dma_data = &s3c_ac97_pcm_out; + else + cpu_dai->dma_data = &s3c_ac97_pcm_in; + + return 0; +} + +static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; + else + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; + else + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + else + cpu_dai->dma_data = &s3c_ac97_mic_in; + + return 0; +} + +static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static struct snd_soc_dai_ops s3c_ac97_dai_ops = { + .hw_params = s3c_ac97_hw_params, + .trigger = s3c_ac97_trigger, +}; + +static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { + .hw_params = s3c_ac97_hw_mic_params, + .trigger = s3c_ac97_mic_trigger, +}; + +struct snd_soc_dai s3c_ac97_dai[] = { + [S3C_AC97_DAI_PCM] = { + .name = "s3c-ac97", + .id = S3C_AC97_DAI_PCM, + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_dai_ops, + }, + [S3C_AC97_DAI_MIC] = { + .name = "s3c-ac97-mic", + .id = S3C_AC97_DAI_MIC, + .ac97_control = 1, + .capture = { + .stream_name = "AC97 Mic Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_mic_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(s3c_ac97_dai); + +static __devinit int s3c_ac97_probe(struct platform_device *pdev) +{ + struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; + struct s3c_audio_pdata *ac97_pdata; + int ret; + + ac97_pdata = pdev->dev.platform_data; + if (!ac97_pdata || !ac97_pdata->cfg_gpio) { + dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); + return -EINVAL; + } + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); + return -ENXIO; + } + + dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); + if (!dmamic_res) { + dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!irq_res) { + dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); + return -ENXIO; + } + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "s3c-ac97")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + return -EBUSY; + } + + s3c_ac97_pcm_out.channel = dmatx_res->start; + s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_pcm_in.channel = dmarx_res->start; + s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_mic_in.channel = dmamic_res->start; + s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; + + init_completion(&s3c_ac97.done); + mutex_init(&s3c_ac97.lock); + + s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); + if (s3c_ac97.regs == NULL) { + dev_err(&pdev->dev, "Unable to ioremap register region\n"); + ret = -ENXIO; + goto err1; + } + + s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + if (IS_ERR(s3c_ac97.ac97_clk)) { + dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); + ret = -ENODEV; + goto err2; + } + clk_enable(s3c_ac97.ac97_clk); + + if (ac97_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err3; + } + + ret = request_irq(irq_res->start, s3c_ac97_irq, + IRQF_DISABLED, "AC97", NULL); + if (ret < 0) { + printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); + goto err4; + } + + s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; + s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; + + ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + if (ret) + goto err5; + + return 0; + +err5: + free_irq(irq_res->start, NULL); +err4: +err3: + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); +err2: + iounmap(s3c_ac97.regs); +err1: + release_mem_region(mem_res->start, resource_size(mem_res)); + + return ret; +} + +static __devexit int s3c_ac97_remove(struct platform_device *pdev) +{ + struct resource *mem_res, *irq_res; + + snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (irq_res) + free_irq(irq_res->start, NULL); + + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); + + iounmap(s3c_ac97.regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (mem_res) + release_mem_region(mem_res->start, resource_size(mem_res)); + + return 0; +} + +static struct platform_driver s3c_ac97_driver = { + .probe = s3c_ac97_probe, + .remove = s3c_ac97_remove, + .driver = { + .name = "s3c-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_ac97_init(void) +{ + return platform_driver_register(&s3c_ac97_driver); +} +module_init(s3c_ac97_init); + +static void __exit s3c_ac97_exit(void) +{ + platform_driver_unregister(&s3c_ac97_driver); +} +module_exit(s3c_ac97_exit); + +MODULE_AUTHOR("Jaswinder Singh, "); +MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-ac97.h b/sound/soc/s3c24xx/s3c-ac97.h new file mode 100644 index 0000000..2781983 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-ac97.h @@ -0,0 +1,23 @@ +/* sound/soc/s3c24xx/s3c-ac97.h + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.h + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __S3C_AC97_H_ +#define __S3C_AC97_H_ + +#define S3C_AC97_DAI_PCM 0 +#define S3C_AC97_DAI_MIC 1 + +extern struct snd_soc_dai s3c_ac97_dai[]; + +#endif /* __S3C_AC97_H_ */ -- cgit v0.10.2 From ff6e64dabf66b8e4e7def21857320085fc68db6b Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 14:59:19 +0900 Subject: ASoC: AC97: SMDK: Add wm9713 machine driver This patch adds the common machine driver for SMDKs that have a WM9713 codec attched to the AC97 controller. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index ad3690e..d1c6f93 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -115,3 +115,11 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_S3C24XX_SOC_I2S select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC + +config SND_SOC_SMDK_WM9713 + tristate "SoC AC97 Audio support for SMDK with WM9713" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM9713 + select SND_S3C_SOC_AC97 + help + Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b7411bd..1117678 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o +snd-soc-smdk-wm9713-objs := smdk_wm9713.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -39,3 +40,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o +obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c new file mode 100644 index 0000000..7dd933f --- /dev/null +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -0,0 +1,97 @@ +/* + * smdk_wm9713.c -- SoC audio for SMDK + * + * Copyright 2010 Samsung Electronics Co. Ltd. + * Author: Jaswinder Singh Brar + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation; either version 2 of the + * License, or (at your option) any later version. + * + */ + +#include +#include +#include + +#include "../codecs/wm9713.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk; + +/* + Playback (HeadPhone):- + Headphone Playback Switch - On + $ amixer cset numid=4 1 + + Right Headphone Out Mux - Headphone + $ amixer cset numid=92 2 + Left Headphone Out Mux - Headphone + $ amixer cset numid=93 2 + + Right HP Mixer PCM Playback Switch - On + $ amixer cset numid=75 1 + Left HP Mixer PCM Playback Switch - On + $ amixer cset numid=81 1 + + Capture (LineIn):- + Right Capture Source - Line + $ amixer cset numid=86 2 + Left Capture Source - Line + $ amixer cset numid=87 2 +*/ + +static struct snd_soc_dai_link smdk_dai = { + .name = "AC97", + .stream_name = "AC97 PCM", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], +}; + +static struct snd_soc_card smdk = { + .name = "SMDK", + .platform = &s3c24xx_soc_platform, + .dai_link = &smdk_dai, + .num_links = 1, +}; + +static struct snd_soc_device smdk_snd_ac97_devdata = { + .card = &smdk, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *smdk_snd_ac97_device; + +static int __init smdk_init(void) +{ + int ret; + + smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk_snd_ac97_device, + &smdk_snd_ac97_devdata); + smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; + + ret = platform_device_add(smdk_snd_ac97_device); + if (ret) + platform_device_put(smdk_snd_ac97_device); + + return ret; +} + +static void __exit smdk_exit(void) +{ + platform_device_unregister(smdk_snd_ac97_device); +} + +module_init(smdk_init); +module_exit(smdk_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar@samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From 1ec2963a8cd5fbc5f49dfa20c94229f1b46d1968 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 15:01:03 +0900 Subject: ASoC: AC97: SMDK2443: Switch to s3c-ac97.c Switch to use s3c-ac97.c AC97 controller driver. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index d1c6f93..8b62798 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -75,8 +75,10 @@ config SND_S3C64XX_SOC_WM8580 config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 - select SND_S3C2443_SOC_AC97 + select S3C2410_DMA + select AC97_BUS select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 help Say Y if you want to add support for SoC audio on smdk2443 with the WM9710. diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index 12b783b1..3622588 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -21,7 +21,7 @@ #include "../codecs/ac97.h" #include "s3c-dma.h" -#include "s3c24xx-ac97.h" +#include "s3c-ac97.h" static struct snd_soc_card smdk2443; @@ -29,7 +29,7 @@ static struct snd_soc_dai_link smdk2443_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai = &s3c2443_ac97_dai[0], + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], .codec_dai = &ac97_dai, }, }; -- cgit v0.10.2 From c67d90ffd43a6cf18def21a0de7db56504d78295 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 15:02:04 +0900 Subject: ASoC: AC97: LN2440SBC: Switch to s3c-ac97.c Switch to use s3c-ac97.c AC97 controller driver. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 8b62798..69d143e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -29,9 +29,6 @@ config SND_S3C_SOC_PCM config SND_S3C2443_SOC_AC97 tristate - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_BUS config SND_S3C_SOC_AC97 tristate @@ -86,8 +83,10 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710 config SND_S3C24XX_SOC_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C2443_SOC_AC97 + select S3C2410_DMA + select AC97_BUS select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 help Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650. diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index d00d359..ffa954f 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -25,7 +25,7 @@ #include "../codecs/ac97.h" #include "s3c-dma.h" -#include "s3c24xx-ac97.h" +#include "s3c-ac97.h" static struct snd_soc_card ln2440sbc; @@ -33,7 +33,7 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai = &s3c2443_ac97_dai[0], + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], .codec_dai = &ac97_dai, }, }; -- cgit v0.10.2 From 7beba4d50d5f70c3851f608927882959d532671c Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Wed, 27 Jan 2010 15:04:36 +0900 Subject: ASoC: AC97: S3C2443: Remove unused driver Since, we have generic AC97 controller driver and all the machines have moved to that, there is no need for old s3c2443-ac97.c to exist. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 69d143e..15fe57e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -27,9 +27,6 @@ config SND_S3C64XX_SOC_I2S config SND_S3C_SOC_PCM tristate -config SND_S3C2443_SOC_AC97 - tristate - config SND_S3C_SOC_AC97 tristate select SND_SOC_AC97_BUS diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 1117678..df071a3 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -3,14 +3,12 @@ snd-soc-s3c24xx-objs := s3c-dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o -snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-ac97-objs := s3c-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o -obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c deleted file mode 100644 index 0191e3a..0000000 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ /dev/null @@ -1,432 +0,0 @@ -/* - * s3c2443-ac97.c -- ALSA Soc Audio Layer - * - * (c) 2007 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Copyright (C) 2005, Sean Choi - * All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include -#include - -#include "s3c-dma.h" -#include "s3c24xx-ac97.h" - -struct s3c24xx_ac97_info { - void __iomem *regs; - struct clk *ac97_clk; -}; -static struct s3c24xx_ac97_info s3c24xx_ac97; - -static DECLARE_COMPLETION(ac97_completion); -static u32 codec_ready; -static DEFINE_MUTEX(ac97_mutex); - -static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - u32 ac_glbctrl; - u32 ac_codec_cmd; - u32 stat, addr, data; - - mutex_lock(&ac97_mutex); - - codec_ready = S3C_AC97_GLBSTAT_CODECREADY; - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - wait_for_completion(&ac97_completion); - - stat = readl(s3c24xx_ac97.regs + S3C_AC97_STAT); - addr = (stat >> 16) & 0x7f; - data = (stat & 0xffff); - - if (addr != reg) - printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," - " rep addr = %02x\n", reg, addr); - - mutex_unlock(&ac97_mutex); - - return (unsigned short)data; -} - -static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - u32 ac_glbctrl; - u32 ac_codec_cmd; - - mutex_lock(&ac97_mutex); - - codec_ready = S3C_AC97_GLBSTAT_CODECREADY; - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - wait_for_completion(&ac97_completion); - - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - mutex_unlock(&ac97_mutex); - -} - -static void s3c2443_ac97_warm_reset(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_WARMRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); -} - -static void s3c2443_ac97_cold_reset(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA | - S3C_AC97_GLBCTRL_PCMINTM_DMA | S3C_AC97_GLBCTRL_MICINTM_DMA; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); -} - -static irqreturn_t s3c2443_ac97_irq(int irq, void *dev_id) -{ - int status; - u32 ac_glbctrl; - - status = readl(s3c24xx_ac97.regs + S3C_AC97_GLBSTAT) & codec_ready; - - if (status) { - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - complete(&ac97_completion); - } - return IRQ_HANDLED; -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = s3c2443_ac97_read, - .write = s3c2443_ac97_write, - .warm_reset = s3c2443_ac97_warm_reset, - .reset = s3c2443_ac97_cold_reset, -}; - -static struct s3c2410_dma_client s3c2443_dma_client_out = { - .name = "AC97 PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2443_dma_client_in = { - .name = "AC97 PCM Stereo in" -}; - -static struct s3c2410_dma_client s3c2443_dma_client_micin = { - .name = "AC97 Mic Mono in" -}; - -static struct s3c_dma_params s3c2443_ac97_pcm_stereo_out = { - .client = &s3c2443_dma_client_out, - .channel = DMACH_PCM_OUT, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2443_ac97_pcm_stereo_in = { - .client = &s3c2443_dma_client_in, - .channel = DMACH_PCM_IN, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2443_ac97_mic_mono_in = { - .client = &s3c2443_dma_client_micin, - .channel = DMACH_MIC_IN, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, - .dma_size = 4, -}; - -static int s3c2443_ac97_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - int ret; - u32 ac_glbctrl; - - s3c24xx_ac97.regs = ioremap(S3C2440_PA_AC97, 0x100); - if (s3c24xx_ac97.regs == NULL) - return -ENXIO; - - s3c24xx_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); - if (s3c24xx_ac97.ac97_clk == NULL) { - printk(KERN_ERR "s3c2443-ac97 failed to get ac97_clock\n"); - iounmap(s3c24xx_ac97.regs); - return -ENODEV; - } - clk_enable(s3c24xx_ac97.ac97_clk); - - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2443_GPE0_AC_nRESET); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2443_GPE1_AC_SYNC); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2443_GPE2_AC_BITCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2443_GPE3_AC_SDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2443_GPE4_AC_SDO); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - ret = request_irq(IRQ_S3C244x_AC97, s3c2443_ac97_irq, - IRQF_DISABLED, "AC97", NULL); - if (ret < 0) { - printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n"); - clk_disable(s3c24xx_ac97.ac97_clk); - clk_put(s3c24xx_ac97.ac97_clk); - iounmap(s3c24xx_ac97.regs); - } - return ret; -} - -static void s3c2443_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - free_irq(IRQ_S3C244x_AC97, NULL); - clk_disable(s3c24xx_ac97.ac97_clk); - clk_put(s3c24xx_ac97.ac97_clk); - iounmap(s3c24xx_ac97.regs); -} - -static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; - else - cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_in; - - return 0; -} - -static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - else - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - else - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; - break; - } - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - cpu_dai->dma_data = &s3c2443_ac97_mic_mono_in; - - return 0; -} - -static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - } - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -#define s3c2443_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) - -static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = { - .hw_params = s3c2443_ac97_hw_params, - .trigger = s3c2443_ac97_trigger, -}; - -static struct snd_soc_dai_ops s3c2443_ac97_mic_dai_ops = { - .hw_params = s3c2443_ac97_hw_mic_params, - .trigger = s3c2443_ac97_mic_trigger, -}; - -struct snd_soc_dai s3c2443_ac97_dai[] = { -{ - .name = "s3c2443-ac97", - .id = 0, - .ac97_control = 1, - .probe = s3c2443_ac97_probe, - .remove = s3c2443_ac97_remove, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c2443_ac97_dai_ops, -}, -{ - .name = "pxa2xx-ac97-mic", - .id = 1, - .ac97_control = 1, - .capture = { - .stream_name = "AC97 Mic Capture", - .channels_min = 1, - .channels_max = 1, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c2443_ac97_mic_dai_ops, -}, -}; -EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int __init s3c2443_ac97_init(void) -{ - return snd_soc_register_dais(s3c2443_ac97_dai, - ARRAY_SIZE(s3c2443_ac97_dai)); -} -module_init(s3c2443_ac97_init); - -static void __exit s3c2443_ac97_exit(void) -{ - snd_soc_unregister_dais(s3c2443_ac97_dai, - ARRAY_SIZE(s3c2443_ac97_dai)); -} -module_exit(s3c2443_ac97_exit); - - -MODULE_AUTHOR("Graeme Gregory"); -MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h deleted file mode 100644 index e96f941..0000000 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ /dev/null @@ -1,25 +0,0 @@ -/* - * s3c24xx-ac97.c -- ALSA Soc Audio Layer - * - * (c) 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Revision history - * 10th Nov 2006 Initial version. - */ - -#ifndef S3C24XXAC97_H_ -#define S3C24XXAC97_H_ - -#define AC_CMD_ADDR(x) (x << 16) -#define AC_CMD_DATA(x) (x & 0xffff) - -extern struct snd_soc_dai s3c2443_ac97_dai[]; - -#endif /*S3C24XXAC97_H_*/ -- cgit v0.10.2 From 583b2be626b047eeb4f9a26721e38fe4992b2d02 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Jan 2010 20:54:13 +0000 Subject: ASoC: Note jumper settings for smdk_wm9713 driver on SMDK6410 The board supports both GPIO sets for the AC97 bus and the analogue outputs can be switched between this and the WM8580 so add some comments saying what the setup the standard kernel expects is. Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 7dd933f..6fa2c9d 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -22,6 +22,12 @@ static struct snd_soc_card smdk; /* + * Default CFG switch settings to use this driver: + * + * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + */ + +/* Playback (HeadPhone):- Headphone Playback Switch - On $ amixer cset numid=4 1 -- cgit v0.10.2 From 0d34e91596ef537c2893a031f0483014bb82adf3 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 27 Jan 2010 18:56:23 +0100 Subject: ASoC: add a WM8978 codec driver The WM8978 codec from Wolfson Microelectronics is very similar to wm8974, but is stereo and also has some differences in pin configuration and internal signal routing. This driver is based on wm8974 and takes the differences into account. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 62ff26a..0aad72f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8974 if I2C + select SND_SOC_WM8978 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C @@ -230,6 +231,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8974 tristate +config SND_SOC_WM8978 + tristate + config SND_SOC_WM8988 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ea98354..fbd290e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -44,6 +44,7 @@ snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8974-objs := wm8974.o +snd-soc-wm8978-objs := wm8978.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8993-objs := wm8993.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o +obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c new file mode 100644 index 0000000..d9d4e9d --- /dev/null +++ b/sound/soc/codecs/wm8978.c @@ -0,0 +1,1124 @@ +/* + * wm8978.c -- WM8978 ALSA SoC Audio Codec driver + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * Copyright (C) 2007 Carlos Munoz + * Copyright 2006-2009 Wolfson Microelectronics PLC. + * Based on wm8974 and wm8990 by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8978.h" + +static struct snd_soc_codec *wm8978_codec; + +/* wm8978 register cache. Note that register 0 is not included in the cache. */ +static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x00...0x03 */ + 0x0050, 0x0000, 0x0140, 0x0000, /* 0x04...0x07 */ + 0x0000, 0x0000, 0x0000, 0x00ff, /* 0x08...0x0b */ + 0x00ff, 0x0000, 0x0100, 0x00ff, /* 0x0c...0x0f */ + 0x00ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ + 0x002c, 0x002c, 0x002c, 0x0000, /* 0x14...0x17 */ + 0x0032, 0x0000, 0x0000, 0x0000, /* 0x18...0x1b */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x1c...0x1f */ + 0x0038, 0x000b, 0x0032, 0x0000, /* 0x20...0x23 */ + 0x0008, 0x000c, 0x0093, 0x00e9, /* 0x24...0x27 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x28...0x2b */ + 0x0033, 0x0010, 0x0010, 0x0100, /* 0x2c...0x2f */ + 0x0100, 0x0002, 0x0001, 0x0001, /* 0x30...0x33 */ + 0x0039, 0x0039, 0x0039, 0x0039, /* 0x34...0x37 */ + 0x0001, 0x0001, /* 0x38...0x3b */ +}; + +/* codec private data */ +struct wm8978_priv { + struct snd_soc_codec codec; + unsigned int f_pllout; + unsigned int f_mclk; + unsigned int f_256fs; + unsigned int f_opclk; + enum wm8978_sysclk_src sysclk; + u16 reg_cache[WM8978_CACHEREGNUM]; +}; + +static const char *wm8978_companding[] = {"Off", "NC", "u-law", "A-law"}; +static const char *wm8978_eqmode[] = {"Capture", "Playback"}; +static const char *wm8978_bw[] = {"Narrow", "Wide"}; +static const char *wm8978_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz"}; +static const char *wm8978_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz"}; +static const char *wm8978_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz"}; +static const char *wm8978_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"}; +static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"}; +static const char *wm8978_alc3[] = {"ALC", "Limiter"}; +static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"}; + +static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); +static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); +static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); +static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); +static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); +static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); +static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); +static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); + +static const struct snd_kcontrol_new wm8978_snd_controls[] = { + + SOC_SINGLE("Digital Loopback Switch", + WM8978_COMPANDING_CONTROL, 0, 1, 0), + + SOC_ENUM("ADC Companding", adc_compand), + SOC_ENUM("DAC Companding", dac_compand), + + SOC_DOUBLE("DAC Inversion Switch", WM8978_DAC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("PCM Volume", + WM8978_LEFT_DAC_DIGITAL_VOLUME, WM8978_RIGHT_DAC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", WM8978_ADC_CONTROL, 4, 7, 0), + SOC_DOUBLE("ADC Inversion Switch", WM8978_ADC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Volume", + WM8978_LEFT_ADC_DIGITAL_VOLUME, WM8978_RIGHT_ADC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_ENUM("Equaliser Function", eqmode), + SOC_ENUM("EQ1 Cut Off", eq1), + SOC_SINGLE_TLV("EQ1 Volume", WM8978_EQ1, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ2 Bandwith", eq2bw), + SOC_ENUM("EQ2 Cut Off", eq2), + SOC_SINGLE_TLV("EQ2 Volume", WM8978_EQ2, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ3 Bandwith", eq3bw), + SOC_ENUM("EQ3 Cut Off", eq3), + SOC_SINGLE_TLV("EQ3 Volume", WM8978_EQ3, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ4 Bandwith", eq4bw), + SOC_ENUM("EQ4 Cut Off", eq4), + SOC_SINGLE_TLV("EQ4 Volume", WM8978_EQ4, 0, 24, 1, eq_tlv), + + SOC_ENUM("EQ5 Cut Off", eq5), + SOC_SINGLE_TLV("EQ5 Volume", WM8978_EQ5, 0, 24, 1, eq_tlv), + + SOC_SINGLE("DAC Playback Limiter Switch", + WM8978_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Playback Limiter Decay", + WM8978_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Playback Limiter Attack", + WM8978_DAC_LIMITER_1, 0, 15, 0), + + SOC_SINGLE("DAC Playback Limiter Threshold", + WM8978_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE("DAC Playback Limiter Boost", + WM8978_DAC_LIMITER_2, 0, 15, 0), + + SOC_ENUM("ALC Enable Switch", alc1), + SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), + + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), + + SOC_ENUM("ALC Capture Mode", alc3), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + + SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", + WM8978_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("Capture PGA ZC Switch", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + + /* OUT1 - Headphones */ + SOC_DOUBLE_R("Headphone Playback ZC Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Headphone Playback Volume", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT2 - Speakers */ + SOC_DOUBLE_R("Speaker Playback ZC Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Speaker Playback Volume", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT3/4 - Line Output */ + SOC_DOUBLE_R("Line Playback Switch", + WM8978_OUT3_MIXER_CONTROL, WM8978_OUT4_MIXER_CONTROL, 6, 1, 1), + + /* Mixer #3: Boost (Input) mixer */ + SOC_DOUBLE_R("PGA Boost (+20dB)", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 8, 1, 0), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 0, 7, 0, boost_tlv), + + /* Input PGA volume */ + SOC_DOUBLE_R_TLV("Input PGA Volume", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 0, 63, 0, inpga_tlv), + + /* Headphone */ + SOC_DOUBLE_R("Headphone Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 6, 1, 1), + + /* Speaker */ + SOC_DOUBLE_R("Speaker Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), +}; + +/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ +static const struct snd_kcontrol_new wm8978_left_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_LEFT_MIXER_CONTROL, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8978_right_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 0, 1, 0), +}; + +/* OUT3/OUT4 Mixer not implemented */ + +/* Mixer #2: Input PGA Mute */ +static const struct snd_kcontrol_new wm8978_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", WM8978_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new wm8978_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", WM8978_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8978_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 1, 0), + + /* Mixer #1: OUT1,2 */ + SOC_MIXER_ARRAY("Left Output Mixer", WM8978_POWER_MANAGEMENT_3, + 2, 0, wm8978_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", WM8978_POWER_MANAGEMENT_3, + 3, 0, wm8978_right_out_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", WM8978_POWER_MANAGEMENT_2, + 2, 0, wm8978_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", WM8978_POWER_MANAGEMENT_2, + 3, 0, wm8978_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", WM8978_LEFT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", WM8978_RIGHT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", WM8978_POWER_MANAGEMENT_2, + 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", WM8978_POWER_MANAGEMENT_2, + 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", WM8978_POWER_MANAGEMENT_3, + 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", WM8978_POWER_MANAGEMENT_3, + 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT4 VMID", WM8978_POWER_MANAGEMENT_3, + 8, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8978_POWER_MANAGEMENT_1, 4, 0), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Output mixer */ + {"Right Output Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right Output Mixer", "Aux Playback Switch", "RAUX"}, + {"Right Output Mixer", "Line Bypass Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left Output Mixer", "Aux Playback Switch", "LAUX"}, + {"Left Output Mixer", "Line Bypass Switch", "Left Boost Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, +}; + +static int wm8978_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8978_dapm_widgets, + ARRAY_SIZE(wm8978_dapm_widgets)); + + /* set up the WM8978 audio map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* PLL divisors */ +struct wm8978_pll_div { + u32 k; + u8 n; + u8 div2; +}; + +#define FIXED_PLL_SIZE (1 << 24) + +static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 k_part; + unsigned int k, n_div, n_mod; + + n_div = target / source; + if (n_div < 6) { + source >>= 1; + pll_div->div2 = 1; + n_div = target / source; + } else { + pll_div->div2 = 0; + } + + if (n_div < 6 || n_div > 12) + dev_warn(wm8978_codec->dev, + "WM8978 N value exceeds recommended range! N = %u\n", + n_div); + + pll_div->n = n_div; + n_mod = target - source * n_div; + k_part = FIXED_PLL_SIZE * (long long)n_mod + source / 2; + + do_div(k_part, source); + + k = k_part & 0xFFFFFFFF; + + pll_div->k = k; +} +/* + * Calculate internal frequencies and dividers, according to Figure 40 + * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 + */ +static int wm8978_configure_pll(struct snd_soc_codec *codec) +{ + struct wm8978_priv *wm8978 = codec->private_data; + struct wm8978_pll_div pll_div; + unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk, + f_256fs = wm8978->f_256fs; + unsigned int f2, opclk_div; + + if (!f_mclk) + return -EINVAL; + + if (f_opclk) { + /* + * The user needs OPCLK. Choose OPCLKDIV to put + * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. + * f_opclk = f_mclk * prescale * R / 4 / OPCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors(). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 16 <= f_opclk < f_mclk * 13 / 4. + */ + if (16 * f_opclk < 3 * f_mclk || 4 * f_opclk >= 13 * f_mclk) + return -EINVAL; + + if (4 * f_opclk < 3 * f_mclk) + /* Have to use OPCLKDIV */ + opclk_div = (3 * f_mclk / 4 + f_opclk - 1) / f_opclk; + else + opclk_div = 1; + + dev_dbg(codec->dev, "%s: OPCLKDIV=%d\n", __func__, opclk_div); + + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 0x30, + (opclk_div - 1) << 4); + + wm8978->f_pllout = f_opclk * opclk_div; + } else if (f_256fs) { + /* + * Not using OPCLK, choose R: + * 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12. + * f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors(). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK + * must be 3.781MHz <= f_MCLK <= 32.768MHz + */ + if (48 * f_256fs < 3 * f_mclk || 4 * f_256fs >= 13 * f_mclk) + return -EINVAL; + + /* + * MCLKDIV will be selected in .hw_params(), just choose a + * suitable f_PLLOUT + */ + if (4 * f_256fs < 3 * f_mclk) + /* Will have to use MCLKDIV */ + wm8978->f_pllout = wm8978->f_mclk * 3 / 4; + else + wm8978->f_pllout = f_256fs; + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + } else { + return -EINVAL; + } + + f2 = wm8978->f_pllout * 4; + + dev_dbg(codec->dev, "%s: f_MCLK=%uHz, f_PLLOUT=%uHz\n", __func__, + wm8978->f_mclk, wm8978->f_pllout); + + pll_factors(&pll_div, f2, wm8978->f_mclk); + + dev_dbg(codec->dev, "%s: calculated PLL N=0x%x, K=0x%x, div2=%d\n", + __func__, pll_div.n, pll_div.k, pll_div.div2); + + /* Turn PLL off for configuration... */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + + snd_soc_write(codec, WM8978_PLL_N, (pll_div.div2 << 4) | pll_div.n); + snd_soc_write(codec, WM8978_PLL_K1, pll_div.k >> 18); + snd_soc_write(codec, WM8978_PLL_K2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8978_PLL_K3, pll_div.k & 0x1ff); + + /* ...and on again */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + if (f_opclk) + /* Output PLL (OPCLK) to GPIO1 */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 4); + + return 0; +} + +/* + * Configure WM8978 clock dividers. + */ +static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + switch (div_id) { + case WM8978_OPCLKRATE: + wm8978->f_opclk = div; + + if (wm8978->f_mclk) + ret = wm8978_configure_pll(codec); + break; + case WM8978_MCLKDIV: + if (div & ~0xe0) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); + break; + case WM8978_ADCCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); + break; + case WM8978_DACCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); + break; + case WM8978_BCLKDIV: + if (div & ~0x1c) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x1c, div); + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "%s: ID %d, value %u\n", __func__, div_id, div); + + return ret; +} + +/* + * @freq: when .set_pll() us not used, freq is codec MCLK input frequency + */ +static int wm8978_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + dev_dbg(codec->dev, "%s: ID %d, freq %u\n", __func__, clk_id, freq); + + if (freq) { + wm8978->f_mclk = freq; + + /* Even if MCLK is used for system clock, might have to drive OPCLK */ + if (wm8978->f_opclk) + ret = wm8978_configure_pll(codec); + + /* Our sysclk is fixed to 256 * fs, will configure in .hw_params() */ + + if (!ret) + wm8978->sysclk = clk_id; + } + + if (wm8978->sysclk == WM8978_PLL && (!freq || clk_id == WM8978_MCLK)) { + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + wm8978->sysclk = WM8978_MCLK; + wm8978->f_pllout = 0; + wm8978->f_opclk = 0; + } + + return ret; +} + +/* + * Set ADC and Voice DAC format. + */ +static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + /* + * BCLK polarity mask = 0x100, LRC clock polarity mask = 0x80, + * Data Format mask = 0x18: all will be calculated anew + */ + u16 iface = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x198; + u16 clk = snd_soc_read(codec, WM8978_CLOCKING); + + dev_dbg(codec->dev, "%s\n", __func__); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + clk &= ~1; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x18; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface); + snd_soc_write(codec, WM8978_CLOCKING, clk); + + return 0; +} + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8978_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + /* Word length mask = 0x60 */ + u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; + /* Sampling rate mask = 0xe (for filters) */ + u16 add_ctl = snd_soc_read(codec, WM8978_ADDITIONAL_CONTROL) & ~0xe; + u16 clking = snd_soc_read(codec, WM8978_CLOCKING); + enum wm8978_sysclk_src current_clk_id = clking & 0x100 ? + WM8978_PLL : WM8978_MCLK; + unsigned int f_sel, diff, diff_best = INT_MAX; + int i, best = 0; + + if (!wm8978->f_mclk) + return -EINVAL; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_ctl |= 0x20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_ctl |= 0x40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_ctl |= 0x60; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case 8000: + add_ctl |= 0x5 << 1; + break; + case 11025: + add_ctl |= 0x4 << 1; + break; + case 16000: + add_ctl |= 0x3 << 1; + break; + case 22050: + add_ctl |= 0x2 << 1; + break; + case 32000: + add_ctl |= 0x1 << 1; + break; + case 44100: + case 48000: + break; + } + + /* Sampling rate is known now, can configure the MCLK divider */ + wm8978->f_256fs = params_rate(params) * 256; + + if (wm8978->sysclk == WM8978_MCLK) { + f_sel = wm8978->f_mclk; + } else { + if (!wm8978->f_pllout) { + int ret = wm8978_configure_pll(codec); + if (ret < 0) + return ret; + } + f_sel = wm8978->f_pllout; + } + + /* + * In some cases it is possible to reconfigure PLL to a higher frequency + * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or + * 512 * fs, so, we should be fine. + */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + + if (diff < diff_best) { + diff_best = diff; + best = i; + } + + if (!diff) + break; + } + + if (diff) + dev_warn(codec->dev, "Imprecise clock: %u%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best], + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); + + dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, + params_format(params), params_rate(params), best); + + /* MCLK divisor mask = 0xe0 */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, best << 5); + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface_ctl); + snd_soc_write(codec, WM8978_ADDITIONAL_CONTROL, add_ctl); + + if (wm8978->sysclk != current_clk_id) { + if (wm8978->sysclk == WM8978_PLL) + /* Run CODEC from PLL instead of MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, + 0x100, 0x100); + else + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + } + + return 0; +} + +static int wm8978_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int wm8978_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + power1 |= 1; /* VMID 75k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_STANDBY: + /* bit 3: enable bias, bit 2: enable I/O tie off buffer */ + power1 |= 0xc; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, + power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_OFF: + /* Preserve PLL - OPCLK may be used by someone */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, ~0x20, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1); + + codec->bias_level = level; + return 0; +} + +#define WM8978_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8978_dai_ops = { + .hw_params = wm8978_hw_params, + .digital_mute = wm8978_mute, + .set_fmt = wm8978_set_dai_fmt, + .set_clkdiv = wm8978_set_dai_clkdiv, + .set_sysclk = wm8978_set_dai_sysclk, +}; + +/* Also supports 12kHz */ +struct snd_soc_dai wm8978_dai = { + .name = "WM8978 HiFi", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .ops = &wm8978_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8978_dai); + +static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); + /* Also switch PLL off */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); + /* Put to sleep */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); + + return 0; +} + +static int wm8978_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int i; + u16 *cache = codec->reg_cache; + + /* Wake up the codec */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { + if (i == WM8978_RESET) + continue; + if (cache[i] != wm8978_reg[i]) + snd_soc_write(codec, i, cache[i]); + } + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (wm8978->f_pllout) + /* Switch PLL on */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + return 0; +} + +static int wm8978_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8978_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8978_codec; + codec = wm8978_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8978_snd_controls, + ARRAY_SIZE(wm8978_snd_controls)); + wm8978_add_widgets(codec); + +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8978_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8978 = { + .probe = wm8978_probe, + .remove = wm8978_remove, + .suspend = wm8978_suspend, + .resume = wm8978_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8978); + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + WM8978_LEFT_DAC_DIGITAL_VOLUME, + WM8978_RIGHT_DAC_DIGITAL_VOLUME, + WM8978_LEFT_ADC_DIGITAL_VOLUME, + WM8978_RIGHT_ADC_DIGITAL_VOLUME, + WM8978_LEFT_INP_PGA_CONTROL, + WM8978_RIGHT_INP_PGA_CONTROL, + WM8978_LOUT1_HP_CONTROL, + WM8978_ROUT1_HP_CONTROL, + WM8978_LOUT2_SPK_CONTROL, + WM8978_ROUT2_SPK_CONTROL, +}; + +static __devinit int wm8978_register(struct wm8978_priv *wm8978) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8978->codec; + + if (wm8978_codec) { + dev_err(codec->dev, "Another WM8978 is registered\n"); + return -EINVAL; + } + + /* + * Set default system clock to PLL, it is more precise, this is also the + * default hardware setting + */ + wm8978->sysclk = WM8978_PLL; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8978; + codec->name = "WM8978"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8978_set_bias_level; + codec->dai = &wm8978_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8978_CACHEREGNUM; + codec->reg_cache = &wm8978->reg_cache; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + memcpy(codec->reg_cache, wm8978_reg, sizeof(wm8978_reg)); + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + + /* Reset the codec */ + ret = snd_soc_write(codec, WM8978_RESET, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + wm8978_dai.dev = codec->dev; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8978_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8978_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8978); + return ret; +} + +static __devexit void wm8978_unregister(struct wm8978_priv *wm8978) +{ + wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8978_dai); + snd_soc_unregister_codec(&wm8978->codec); + kfree(wm8978); + wm8978_codec = NULL; +} + +static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8978_priv *wm8978; + struct snd_soc_codec *codec; + + wm8978 = kzalloc(sizeof(struct wm8978_priv), GFP_KERNEL); + if (wm8978 == NULL) + return -ENOMEM; + + codec = &wm8978->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8978); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8978_register(wm8978); +} + +static __devexit int wm8978_i2c_remove(struct i2c_client *client) +{ + struct wm8978_priv *wm8978 = i2c_get_clientdata(client); + wm8978_unregister(wm8978); + return 0; +} + +static const struct i2c_device_id wm8978_i2c_id[] = { + { "wm8978", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); + +static struct i2c_driver wm8978_i2c_driver = { + .driver = { + .name = "WM8978", + .owner = THIS_MODULE, + }, + .probe = wm8978_i2c_probe, + .remove = __devexit_p(wm8978_i2c_remove), + .id_table = wm8978_i2c_id, +}; + +static int __init wm8978_modinit(void) +{ + return i2c_add_driver(&wm8978_i2c_driver); +} +module_init(wm8978_modinit); + +static void __exit wm8978_exit(void) +{ + i2c_del_driver(&wm8978_i2c_driver); +} +module_exit(wm8978_exit); + +MODULE_DESCRIPTION("ASoC WM8978 codec driver"); +MODULE_AUTHOR("Guennadi Liakhovetski "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h new file mode 100644 index 0000000..b58f0bf --- /dev/null +++ b/sound/soc/codecs/wm8978.h @@ -0,0 +1,89 @@ +/* + * wm8978.h -- codec driver for WM8978 + * + * Copyright 2009 Guennadi Liakhovetski + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __WM8978_H__ +#define __WM8978_H__ + +/* + * Register values. + */ +#define WM8978_RESET 0x00 +#define WM8978_POWER_MANAGEMENT_1 0x01 +#define WM8978_POWER_MANAGEMENT_2 0x02 +#define WM8978_POWER_MANAGEMENT_3 0x03 +#define WM8978_AUDIO_INTERFACE 0x04 +#define WM8978_COMPANDING_CONTROL 0x05 +#define WM8978_CLOCKING 0x06 +#define WM8978_ADDITIONAL_CONTROL 0x07 +#define WM8978_GPIO_CONTROL 0x08 +#define WM8978_JACK_DETECT_CONTROL_1 0x09 +#define WM8978_DAC_CONTROL 0x0A +#define WM8978_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8978_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8978_JACK_DETECT_CONTROL_2 0x0D +#define WM8978_ADC_CONTROL 0x0E +#define WM8978_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8978_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8978_EQ1 0x12 +#define WM8978_EQ2 0x13 +#define WM8978_EQ3 0x14 +#define WM8978_EQ4 0x15 +#define WM8978_EQ5 0x16 +#define WM8978_DAC_LIMITER_1 0x18 +#define WM8978_DAC_LIMITER_2 0x19 +#define WM8978_NOTCH_FILTER_1 0x1b +#define WM8978_NOTCH_FILTER_2 0x1c +#define WM8978_NOTCH_FILTER_3 0x1d +#define WM8978_NOTCH_FILTER_4 0x1e +#define WM8978_ALC_CONTROL_1 0x20 +#define WM8978_ALC_CONTROL_2 0x21 +#define WM8978_ALC_CONTROL_3 0x22 +#define WM8978_NOISE_GATE 0x23 +#define WM8978_PLL_N 0x24 +#define WM8978_PLL_K1 0x25 +#define WM8978_PLL_K2 0x26 +#define WM8978_PLL_K3 0x27 +#define WM8978_3D_CONTROL 0x29 +#define WM8978_BEEP_CONTROL 0x2b +#define WM8978_INPUT_CONTROL 0x2c +#define WM8978_LEFT_INP_PGA_CONTROL 0x2d +#define WM8978_RIGHT_INP_PGA_CONTROL 0x2e +#define WM8978_LEFT_ADC_BOOST_CONTROL 0x2f +#define WM8978_RIGHT_ADC_BOOST_CONTROL 0x30 +#define WM8978_OUTPUT_CONTROL 0x31 +#define WM8978_LEFT_MIXER_CONTROL 0x32 +#define WM8978_RIGHT_MIXER_CONTROL 0x33 +#define WM8978_LOUT1_HP_CONTROL 0x34 +#define WM8978_ROUT1_HP_CONTROL 0x35 +#define WM8978_LOUT2_SPK_CONTROL 0x36 +#define WM8978_ROUT2_SPK_CONTROL 0x37 +#define WM8978_OUT3_MIXER_CONTROL 0x38 +#define WM8978_OUT4_MIXER_CONTROL 0x39 + +#define WM8978_CACHEREGNUM 58 + +/* Clock divider Id's */ +enum wm8978_clk_id { + WM8978_OPCLKRATE, + WM8978_MCLKDIV, + WM8978_ADCCLK, + WM8978_DACCLK, + WM8978_BCLKDIV, +}; + +enum wm8978_sysclk_src { + WM8978_PLL, + WM8978_MCLK +}; + +extern struct snd_soc_dai wm8978_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8978; + +#endif /* __WM8978_H__ */ -- cgit v0.10.2 From b09f3e78ee7bb69171411b75bd9e771fc7f24749 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jan 2010 00:01:53 +0100 Subject: ALSA: hda - Allow override more fields via patch loader Allow the override of vendor-id, subsystem-id, revision-id and chip name via patch loading. Updated the document, too. Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 6325bec..f4dd3bf 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -452,6 +452,33 @@ Similarly, the lines after `[verb]` are parsed as `init_verbs` sysfs entries, and the lines after `[hint]` are parsed as `hints` sysfs entries, respectively. +Another example to override the codec vendor id from 0x12345678 to +0xdeadbeef is like below: +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [vendor_id] + 0xdeadbeef +------------------------------------------------------------------------ + +In the similar way, you can override the codec subsystem_id via +`[subsystem_id]`, the revision id via `[revision_id]` line. +Also, the codec chip name can be rewritten via `[chip_name]` line. +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [subsystem_id] + 0xffff1111 + + [revision_id] + 0x10 + + [chip_name] + My-own NEWS-0002 +------------------------------------------------------------------------ + The hd-audio driver reads the file via request_firmware(). Thus, a patch file has to be located on the appropriate firmware path, typically, /lib/firmware. For example, when you pass the option diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index b36919c..a1fc837 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -625,6 +625,10 @@ enum { LINE_MODE_PINCFG, LINE_MODE_VERB, LINE_MODE_HINT, + LINE_MODE_VENDOR_ID, + LINE_MODE_SUBSYSTEM_ID, + LINE_MODE_REVISION_ID, + LINE_MODE_CHIP_NAME, NUM_LINE_MODES, }; @@ -654,53 +658,71 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus, } /* parse the contents after the other command tags, [pincfg], [verb], - * [hint] and [model] + * [vendor_id], [subsystem_id], [revision_id], [chip_name], [hint] and [model] * just pass to the sysfs helper (only when any codec was specified) */ static void parse_pincfg_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - if (!*codecp) - return; parse_user_pin_configs(*codecp, buf); } static void parse_verb_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - if (!*codecp) - return; parse_init_verbs(*codecp, buf); } static void parse_hint_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - if (!*codecp) - return; parse_hints(*codecp, buf); } static void parse_model_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - if (!*codecp) - return; kfree((*codecp)->modelname); (*codecp)->modelname = kstrdup(buf, GFP_KERNEL); } +static void parse_chip_name_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + kfree((*codecp)->chip_name); + (*codecp)->chip_name = kstrdup(buf, GFP_KERNEL); +} + +#define DEFINE_PARSE_ID_MODE(name) \ +static void parse_##name##_mode(char *buf, struct hda_bus *bus, \ + struct hda_codec **codecp) \ +{ \ + unsigned long val; \ + if (!strict_strtoul(buf, 0, &val)) \ + (*codecp)->name = val; \ +} + +DEFINE_PARSE_ID_MODE(vendor_id); +DEFINE_PARSE_ID_MODE(subsystem_id); +DEFINE_PARSE_ID_MODE(revision_id); + + struct hda_patch_item { const char *tag; void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc); + int need_codec; }; static struct hda_patch_item patch_items[NUM_LINE_MODES] = { - [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode }, - [LINE_MODE_MODEL] = { "[model]", parse_model_mode }, - [LINE_MODE_VERB] = { "[verb]", parse_verb_mode }, - [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode }, - [LINE_MODE_HINT] = { "[hint]", parse_hint_mode }, + [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode, 0 }, + [LINE_MODE_MODEL] = { "[model]", parse_model_mode, 1 }, + [LINE_MODE_VERB] = { "[verb]", parse_verb_mode, 1 }, + [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode, 1 }, + [LINE_MODE_HINT] = { "[hint]", parse_hint_mode, 1 }, + [LINE_MODE_VENDOR_ID] = { "[vendor_id]", parse_vendor_id_mode, 1 }, + [LINE_MODE_SUBSYSTEM_ID] = { "[subsystem_id]", parse_subsystem_id_mode, 1 }, + [LINE_MODE_REVISION_ID] = { "[revision_id]", parse_revision_id_mode, 1 }, + [LINE_MODE_CHIP_NAME] = { "[chip_name]", parse_chip_name_mode, 1 }, }; /* check the line starting with '[' -- change the parser mode accodingly */ @@ -783,7 +805,8 @@ int snd_hda_load_patch(struct hda_bus *bus, const char *patch) continue; if (*buf == '[') line_mode = parse_line_mode(buf, bus); - else if (patch_items[line_mode].parser) + else if (patch_items[line_mode].parser && + (codec || !patch_items[line_mode].need_codec)) patch_items[line_mode].parser(buf, bus, &codec); } release_firmware(fw); -- cgit v0.10.2 From 8fc176d5abb2d92c52df859faac7974b4a1585c1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Jan 2010 13:46:16 +0900 Subject: ASoC: fsi: Add spin lock operation for accessing shared area fsi_master_xxx function should be protected by spin lock, because it are used from both FSI-A and FSI-B. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5f9f269..ebf3588 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -110,6 +110,7 @@ struct fsi_master { struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; + spinlock_t lock; }; /************************************************************************ @@ -168,30 +169,51 @@ static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { + int ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return -1; - return __fsi_reg_write((u32)(master->base + reg), data); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_write((u32)(master->base + reg), data); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } static u32 fsi_master_read(struct fsi_master *master, u32 reg) { + u32 ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return 0; - return __fsi_reg_read((u32)(master->base + reg)); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_read((u32)(master->base + reg)); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } static int fsi_master_mask_set(struct fsi_master *master, u32 reg, u32 mask, u32 data) { + int ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return -1; - return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } /************************************************************************ @@ -929,6 +951,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsib.base = master->base + 0x40; master->fsib.master = master; + spin_lock_init(&master->lock); pm_runtime_enable(&pdev->dev); pm_runtime_resume(&pdev->dev); -- cgit v0.10.2 From c812459396733b42655e0d656763af02e06f97ed Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 28 Jan 2010 15:57:04 +0200 Subject: ASoC: TWL4030: Modify codec default settings Change the legacy default register configuration, which left some internal components on. Now we have either DAPM, or other ways to control these bits, so there is no need to enable them by default. The affected parts: Disable ADCL and ADCR Disable ARXL2 and ARXR2 analog PGA (playback) Disable APLL by default Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 74f0d65..e0106a5 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -64,12 +64,12 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VRXPGA (0x14) */ 0x00, /* REG_VSTPGA (0x15) */ 0x00, /* REG_VRX2ARXPGA (0x16) */ - 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_AVDAC_CTL (0x17) */ 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x4a, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x4a, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ 0x00, /* REG_BTPGA (0x1F) */ @@ -99,7 +99,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ - 0x16, /* REG_APLL_CTL (0x3A) */ + 0x06, /* REG_APLL_CTL (0x3A) */ 0x00, /* REG_DTMF_CTL (0x3B) */ 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ -- cgit v0.10.2 From fb58a2ff300cb3fd6077484ca7d8c6e6f13a0350 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Jan 2010 10:22:45 +0000 Subject: ASoC: Remove version display from WM9713 The version isn't being updated or used, the kernel revision tracking is enough. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index c58aab3..96e46d9 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -28,8 +28,6 @@ #include "wm9713.h" -#define WM9713_VERSION "0.15" - struct wm9713_priv { u32 pll_in; /* PLL input frequency */ }; @@ -1186,8 +1184,6 @@ static int wm9713_soc_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret = 0, reg; - printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); - socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (socdev->card->codec == NULL) -- cgit v0.10.2 From e03a8d2cf663429e2480a8db78b132ee300f79af Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Jan 2010 12:36:07 +0000 Subject: ASoC: Add TLV information and additional volumes to WM9713 Also renames a few things to make volumes and switches match up in alsamixer. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 96e46d9..ceb86b4 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include @@ -113,15 +114,27 @@ SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ }; +static const DECLARE_TLV_DB_SCALE(out_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(main_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(misc_tlv, -1500, 300, 0); +static unsigned int mic_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = { -SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), +SOC_DOUBLE_TLV("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1, out_tlv), SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1), -SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), +SOC_DOUBLE_TLV("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1, + out_tlv), SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1), -SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), -SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), -SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), -SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), +SOC_DOUBLE_TLV("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1, main_tlv), +SOC_DOUBLE_TLV("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1, main_tlv), +SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), +SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv), +SOC_SINGLE_TLV("Mic 1 Preamp Volume", AC97_3D_CONTROL, 10, 3, 0, mic_tlv), +SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 12, 3, 0, mic_tlv), SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), @@ -131,7 +144,7 @@ SOC_ENUM("Capture Volume Steps", wm9713_enum[5]), SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0), SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), -SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), +SOC_SINGLE_TLV("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1, misc_tlv), SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), @@ -152,28 +165,43 @@ SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0), -SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1), +SOC_SINGLE_TLV("Out4 Playback Volume", AC97_MASTER_MONO, 8, 31, 1, out_tlv), SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1), SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0), -SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1), +SOC_SINGLE_TLV("Out3 Playback Volume", AC97_MASTER_MONO, 0, 31, 1, out_tlv), -SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1), +SOC_SINGLE_TLV("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1, main_tlv), SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), +SOC_SINGLE_TLV("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1, out_tlv), -SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), -SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), -SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), +SOC_SINGLE_TLV("Headphone Mixer Beep Playback Volume", AC97_AUX, 12, 7, 1, + misc_tlv), +SOC_SINGLE_TLV("Speaker Mixer Beep Playback Volume", AC97_AUX, 8, 7, 1, + misc_tlv), +SOC_SINGLE_TLV("Mono Mixer Beep Playback Volume", AC97_AUX, 4, 7, 1, misc_tlv), -SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), +SOC_SINGLE_TLV("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1, + misc_tlv), SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1), +SOC_SINGLE_TLV("Headphone Mixer Aux Playback Volume", AC97_REC_SEL, 12, 7, 1, + misc_tlv), + +SOC_SINGLE_TLV("Speaker Mixer Voice Playback Volume", AC97_PCM, 8, 7, 1, + misc_tlv), +SOC_SINGLE_TLV("Speaker Mixer Aux Playback Volume", AC97_REC_SEL, 8, 7, 1, + misc_tlv), + +SOC_SINGLE_TLV("Mono Mixer Voice Playback Volume", AC97_PCM, 4, 7, 1, + misc_tlv), +SOC_SINGLE_TLV("Mono Mixer Aux Playback Volume", AC97_REC_SEL, 4, 7, 1, + misc_tlv), + SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1), SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1), -SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1), SOC_ENUM("Bass Control", wm9713_enum[16]), SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), -- cgit v0.10.2 From 2718625fba1e07bf2ce8a752036658737c1f76a7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Jan 2010 12:36:29 +0000 Subject: ASoC: Set codec->dev for AC97 devices Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9085b40..ca89c78 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1368,6 +1368,7 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, codec->ac97->bus->ops = ops; codec->ac97->num = num; + codec->dev = &codec->ac97->dev; mutex_unlock(&codec->mutex); return 0; } -- cgit v0.10.2 From 7b36ea967cc5b5088a57fe225f1f72a3c160058b Mon Sep 17 00:00:00 2001 From: Wei Ni Date: Thu, 28 Jan 2010 16:13:07 +0800 Subject: ALSA: hda - Change the AZX_MAX_PCMS to 10 In hda_codec.c, it has define "[HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 },", it support up to device 9 for HDMI. But in hda_intel.c, it only define AZX_MAX_PCMS as 8. So if it have 4 hdmi codecs, when run azx_attach_pcm_stream(), it will show error "Invalid PCM device number 8", and "... number 9", and return "-EINVAL". We should change the AZX_MAX_PCMS to 10. Signed-off-by: Wei Ni Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6eeefda..170126c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -261,7 +261,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* max buffer size - no h/w limit, you can increase as you like */ #define AZX_MAX_BUF_SIZE (1024*1024*1024) /* max number of PCM devics per card */ -#define AZX_MAX_PCMS 8 +#define AZX_MAX_PCMS 10 /* RIRB int mask: overrun[2], response[0] */ #define RIRB_INT_RESPONSE 0x01 -- cgit v0.10.2 From c89362225152fc6f2247f65371bfe3ccced3203b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jan 2010 17:08:53 +0100 Subject: ALSA: hda - Define max number of PCM devices in hda_codec.h Define the constant rather in the common header file. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 26ceace..98767df 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3275,6 +3275,8 @@ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { /* * get the empty PCM device number to assign + * + * note the max device number is limited by HDA_MAX_PCMS, currently 10 */ static int get_empty_pcm_device(struct hda_bus *bus, int type) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0c8f05c..b75da47 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -527,6 +527,9 @@ enum { /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f +/* max number of PCM devics per card */ +#define HDA_MAX_PCMS 10 + /* * generic arrays */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170126c..12230a2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -260,8 +260,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_MAX_FRAG 32 /* max buffer size - no h/w limit, you can increase as you like */ #define AZX_MAX_BUF_SIZE (1024*1024*1024) -/* max number of PCM devics per card */ -#define AZX_MAX_PCMS 10 /* RIRB int mask: overrun[2], response[0] */ #define RIRB_INT_RESPONSE 0x01 @@ -409,7 +407,7 @@ struct azx { struct azx_dev *azx_dev; /* PCM */ - struct snd_pcm *pcm[AZX_MAX_PCMS]; + struct snd_pcm *pcm[HDA_MAX_PCMS]; /* HD codec */ unsigned short codec_mask; @@ -1336,7 +1334,7 @@ static void azx_bus_reset(struct hda_bus *bus) if (chip->initialized) { int i; - for (i = 0; i < AZX_MAX_PCMS; i++) + for (i = 0; i < HDA_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus); snd_hda_resume(chip->bus); @@ -1966,7 +1964,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, int pcm_dev = cpcm->device; int s, err; - if (pcm_dev >= AZX_MAX_PCMS) { + if (pcm_dev >= HDA_MAX_PCMS) { snd_printk(KERN_ERR SFX "Invalid PCM device number %d\n", pcm_dev); return -EINVAL; @@ -2122,7 +2120,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_clear_irq_pending(chip); - for (i = 0; i < AZX_MAX_PCMS; i++) + for (i = 0; i < HDA_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) snd_hda_suspend(chip->bus); -- cgit v0.10.2 From 30ed7ed11cb88fd56d821a67b9aab1e0d50fb626 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jan 2010 17:11:45 +0100 Subject: ALSA: hda - Fix index of HP Compaq F700 mic amp The amp used for the mic input on HP Compaq F700 with Cxt5051 codec has no multiple inputs, thus its index should be 0 instead of 1. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9077e41..745e359 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1832,7 +1832,7 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { static struct hda_verb cxt5051_f700_init_verbs[] = { /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x03}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, -- cgit v0.10.2 From e108c7b79e91b45a3f04762c44fd404a5d9be069 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Thu, 28 Jan 2010 19:21:07 +0100 Subject: ALSA: hda - Add mute LED check for HP laptops with IDT 92HD83xxx codec This patch adds HP mute LED support for IDT 92HD81/3 family of the codecs. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dbffb5b..cb9802f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5332,6 +5332,11 @@ again: if (spec->board_config == STAC_92HD83XXX_HP) spec->gpio_led = 0x01; + if (find_mute_led_gpio(codec)) + snd_printd("mute LED gpio %d polarity %d\n", + spec->gpio_led, + spec->gpio_led_polarity); + #ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { spec->gpio_mask |= spec->gpio_led; -- cgit v0.10.2 From 36706005d90642bccabfaacbb24d135155e984a8 Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Fri, 29 Jan 2010 12:05:51 +0100 Subject: ALSA: hda - Add support for IDT 92HD88 family codecs Signed-off-by: Charles Chin Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index cb9802f..9694675 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -568,6 +568,11 @@ static hda_nid_t stac92hd83xxx_pin_nids[10] = { 0x0f, 0x10, 0x11, 0x1f, 0x20, }; +static hda_nid_t stac92hd88xxx_pin_nids[10] = { + 0x0a, 0x0b, 0x0c, 0x0d, + 0x0f, 0x11, 0x1f, 0x20, +}; + #define STAC92HD71BXX_NUM_PINS 13 static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x00, @@ -2873,6 +2878,13 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) conn_len = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); + /* 92HD88: trace back up the link of nids to find the DAC */ + while (conn_len == 1 && (get_wcaps_type(get_wcaps(codec, conn[0])) + != AC_WID_AUD_OUT)) { + nid = conn[0]; + conn_len = snd_hda_get_connections(codec, nid, conn, + HDA_MAX_CONNECTIONS); + } for (j = 0; j < conn_len; j++) { wcaps = get_wcaps(codec, conn[j]); wtype = get_wcaps_type(wcaps); @@ -5318,6 +5330,16 @@ again: stac92hd83xxx_brd_tbl[spec->board_config]); switch (codec->vendor_id) { + case 0x111d7666: + case 0x111d7667: + case 0x111d7668: + case 0x111d7669: + spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); + spec->pin_nids = stac92hd88xxx_pin_nids; + spec->mono_nid = 0; + spec->digbeep_nid = 0; + spec->num_pwrs = 0; + break; case 0x111d7604: case 0x111d7605: case 0x111d76d5: @@ -6243,6 +6265,10 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7667, .name = "92HD88B1", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7668, .name = "92HD88B2", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7669, .name = "92HD88B4", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, -- cgit v0.10.2 From 9e9d04c05fd01018da35fa1daa9bda844cac6162 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Fri, 29 Jan 2010 10:57:07 +0900 Subject: ASoC: AC97: SMDK-WM9713: Convert notes from cset to sset It's more robust when references are provided in control names rather than numid. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 6fa2c9d..24fd39f 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -29,24 +29,15 @@ static struct snd_soc_card smdk; /* Playback (HeadPhone):- - Headphone Playback Switch - On - $ amixer cset numid=4 1 - - Right Headphone Out Mux - Headphone - $ amixer cset numid=92 2 - Left Headphone Out Mux - Headphone - $ amixer cset numid=93 2 - - Right HP Mixer PCM Playback Switch - On - $ amixer cset numid=75 1 - Left HP Mixer PCM Playback Switch - On - $ amixer cset numid=81 1 + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute Capture (LineIn):- - Right Capture Source - Line - $ amixer cset numid=86 2 - Left Capture Source - Line - $ amixer cset numid=87 2 + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' */ static struct snd_soc_dai_link smdk_dai = { -- cgit v0.10.2 From 9f5b64b767203131a7a3a280859e70d4413c9672 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 27 Jan 2010 12:15:00 +0100 Subject: ASoC: add support for the sh7722 Migo-R board Add support for audio on sh7722-based Migo-R boards, using SIU and wm8978 codec, recording via external microphone and playback via headphones are implemented. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 3f1cd55..a86696b 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -61,4 +61,12 @@ config SND_FSI_DA7210 This option enables generic sound support for the FSI - DA7210 unit +config SND_SIU_MIGOR + tristate "SIU sound support on Migo-R" + depends on SH_MIGOR + select SND_SOC_SH4_SIU + select SND_SOC_WM8978 + help + This option enables sound support for the SH7722 Migo-R board + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 5a97d25..8a5a192 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -16,7 +16,9 @@ obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o snd-soc-fsi-da7210-objs := fsi-da7210.o +snd-soc-migor-objs := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o +obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c new file mode 100644 index 0000000..3ccd9b3 --- /dev/null +++ b/sound/soc/sh/migor.c @@ -0,0 +1,222 @@ +/* + * ALSA SoC driver for Migo-R + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include + +#include + +#include +#include +#include +#include + +#include "../codecs/wm8978.h" +#include "siu.h" + +/* Default 8000Hz sampling frequency */ +static unsigned long codec_freq = 8000 * 512; + +static unsigned int use_count; + +/* External clock, sourced from the codec at the SIUMCKB pin */ +static unsigned long siumckb_recalc(struct clk *clk) +{ + return codec_freq; +} + +static struct clk_ops siumckb_clk_ops = { + .recalc = siumckb_recalc, +}; + +static struct clk siumckb_clk = { + .name = "siumckb_clk", + .id = -1, + .ops = &siumckb_clk_ops, + .rate = 0, /* initialised at run-time */ +}; + +static int migor_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + unsigned int rate = params_rate(params); + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 13000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(rtd->dai->cpu_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + codec_freq = rate * 512; + /* + * This propagates the parent frequency change to children and + * recalculates the frequency table + */ + clk_set_rate(&siumckb_clk, codec_freq); + dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq); + + ret = snd_soc_dai_set_sysclk(rtd->dai->cpu_dai, SIU_CLKB_EXT, + codec_freq / 2, SND_SOC_CLOCK_IN); + + if (!ret) + use_count++; + + return ret; +} + +static int migor_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + if (use_count) { + use_count--; + + if (!use_count) + snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 0, + SND_SOC_CLOCK_IN); + } else { + dev_dbg(codec_dai->dev, "Unbalanced hw_free!\n"); + } + + return 0; +} + +static struct snd_soc_ops migor_dai_ops = { + .hw_params = migor_hw_params, + .hw_free = migor_hw_free, +}; + +static const struct snd_soc_dapm_widget migor_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Onboard Microphone", NULL), + SND_SOC_DAPM_MIC("External Microphone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone output connected to LHP/RHP, enable OUT4 for VMID */ + { "Headphone", NULL, "OUT4 VMID" }, + { "OUT4 VMID", NULL, "LHP" }, + { "OUT4 VMID", NULL, "RHP" }, + + /* On-board microphone */ + { "RMICN", NULL, "Mic Bias" }, + { "RMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Onboard Microphone" }, + + /* External microphone */ + { "LMICN", NULL, "Mic Bias" }, + { "LMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "External Microphone" }, +}; + +static int migor_dai_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, migor_dapm_widgets, + ARRAY_SIZE(migor_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* migor digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link migor_dai = { + .name = "wm8978", + .stream_name = "WM8978", + .cpu_dai = &siu_i2s_dai, + .codec_dai = &wm8978_dai, + .ops = &migor_dai_ops, + .init = migor_dai_init, +}; + +/* migor audio machine driver */ +static struct snd_soc_card snd_soc_migor = { + .name = "Migo-R", + .platform = &siu_platform, + .dai_link = &migor_dai, + .num_links = 1, +}; + +/* migor audio subsystem */ +static struct snd_soc_device migor_snd_devdata = { + .card = &snd_soc_migor, + .codec_dev = &soc_codec_dev_wm8978, +}; + +static struct platform_device *migor_snd_device; + +static int __init migor_init(void) +{ + int ret; + + ret = clk_register(&siumckb_clk); + if (ret < 0) + return ret; + + /* Port number used on this machine: port B */ + migor_snd_device = platform_device_alloc("soc-audio", 1); + if (!migor_snd_device) { + ret = -ENOMEM; + goto epdevalloc; + } + + platform_set_drvdata(migor_snd_device, &migor_snd_devdata); + + migor_snd_devdata.dev = &migor_snd_device->dev; + + ret = platform_device_add(migor_snd_device); + if (ret) + goto epdevadd; + + return 0; + +epdevadd: + platform_device_put(migor_snd_device); +epdevalloc: + clk_unregister(&siumckb_clk); + return ret; +} + +static void __exit migor_exit(void) +{ + clk_unregister(&siumckb_clk); + platform_device_unregister(migor_snd_device); +} + +module_init(migor_init); +module_exit(migor_exit); + +MODULE_AUTHOR("Guennadi Liakhovetski "); +MODULE_DESCRIPTION("ALSA SoC Migor"); +MODULE_LICENSE("GPL v2"); -- cgit v0.10.2 From 640b796f2ca88113bf2fefd380bc807092ce6fa1 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 28 Jan 2010 16:28:55 +0100 Subject: ASoC: remove bogus SLEEP mode from wm8978 driver Tests showed, that bit 6 of the WM8978_POWER_MANAGEMENT_2 register of wm8978 affects codec clocks. Being useless for suspend / resume, it cannot be used in bias-level control either. Remove this bit handling. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index d9d4e9d..8dcebaa8 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -873,8 +873,6 @@ static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); /* Also switch PLL off */ snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); - /* Put to sleep */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); return 0; } @@ -887,9 +885,6 @@ static int wm8978_resume(struct platform_device *pdev) int i; u16 *cache = codec->reg_cache; - /* Wake up the codec */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { if (i == WM8978_RESET) -- cgit v0.10.2 From b2c3e923110f6ca60ccb30cf4a6bda5211454c4f Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 29 Jan 2010 15:31:06 +0100 Subject: ASoC: clean up wm8974 and wm8978 clock divider handling wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8812751..ee637af 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode), SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), + +/* DAC / ADC oversampling */ +SOC_SINGLE("DAC 128x Oversampling Switch", WM8974_DAC, 8, 1, 0), +SOC_SINGLE("ADC 128x Oversampling Switch", WM8974_ADC, 8, 1, 0), }; /* Speaker Output Mixer */ @@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; snd_soc_write(codec, WM8974_CLOCK, reg | div); break; - case WM8974_ADCCLK: - reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; - snd_soc_write(codec, WM8974_ADC, reg | div); - break; - case WM8974_DACCLK: - reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; - snd_soc_write(codec, WM8974_DAC, reg | div); - break; case WM8974_BCLKDIV: reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; snd_soc_write(codec, WM8974_CLOCK, reg | div); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h index 98de956..896a7f0 100644 --- a/sound/soc/codecs/wm8974.h +++ b/sound/soc/codecs/wm8974.h @@ -57,17 +57,7 @@ /* Clock divider Id's */ #define WM8974_OPCLKDIV 0 #define WM8974_MCLKDIV 1 -#define WM8974_ADCCLK 2 -#define WM8974_DACCLK 3 -#define WM8974_BCLKDIV 4 - -/* DAC clock dividers */ -#define WM8974_DACCLK_F2 (1 << 3) -#define WM8974_DACCLK_F4 (0 << 3) - -/* ADC clock dividers */ -#define WM8974_ADCCLK_F2 (1 << 3) -#define WM8974_ADCCLK_F4 (0 << 3) +#define WM8974_BCLKDIV 2 /* PLL Out dividers */ #define WM8974_OPCLKDIV_1 (0 << 4) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 8dcebaa8..ec2624b 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -210,6 +210,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { /* Speaker */ SOC_DOUBLE_R("Speaker Switch", WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), + + /* DAC / ADC oversampling */ + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ @@ -513,21 +517,6 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, if (wm8978->f_mclk) ret = wm8978_configure_pll(codec); break; - case WM8978_MCLKDIV: - if (div & ~0xe0) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); - break; - case WM8978_ADCCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); - break; - case WM8978_DACCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); - break; case WM8978_BCLKDIV: if (div & ~0x1c) return -EINVAL; diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index b58f0bf..56ec832 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -72,9 +72,6 @@ /* Clock divider Id's */ enum wm8978_clk_id { WM8978_OPCLKRATE, - WM8978_MCLKDIV, - WM8978_ADCCLK, - WM8978_DACCLK, WM8978_BCLKDIV, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 3ccd9b3..b823a5c 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -59,10 +59,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); if (ret < 0) return ret; -- cgit v0.10.2 From a75d7a4cf50d1cee14d8c9aaa2b971232d10f2c1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 1 Feb 2010 13:29:50 +0100 Subject: sound: control: actually allow TLV command access Creating a control with TLV_COMMAND access was not possible because snd_ctl_new1() forgot to include it in the mask of allowable access bits. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/core/control.c b/sound/core/control.c index 268ab74..6a4764d 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -237,8 +237,9 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| + SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); kctl.info = ncontrol->info; kctl.get = ncontrol->get; kctl.put = ncontrol->put; -- cgit v0.10.2 From 6123637fafbf445cc9ce5774dc9516da0b2daa88 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 1 Feb 2010 13:30:56 +0100 Subject: sound: control: fix minimum TLV length Allow TLV blocks that do not have any values; the smallest possible TLV is an empty container or one where the information is only in the tag. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela diff --git a/sound/core/control.c b/sound/core/control.c index 6a4764d..439ce64 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1100,7 +1100,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, if (copy_from_user(&tlv, _tlv, sizeof(tlv))) return -EFAULT; - if (tlv.length < sizeof(unsigned int) * 3) + if (tlv.length < sizeof(unsigned int) * 2) return -EINVAL; down_read(&card->controls_rwsem); kctl = snd_ctl_find_numid(card, tlv.numid); -- cgit v0.10.2 From b0580913797034a1001e867b8b492c75226bf77e Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 29 Jan 2010 14:51:26 +0100 Subject: ASoC: improve MCLKDIV calculation in wm8978, when OPCLK is not used In case, if OPCLK is not used, and PLL is used for driving the codec, the choice of PLL output frequency could result in a needlessly imprecise system clock frequency. Use an iterative process to select a precise configuration. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index ec2624b..28bb59e 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -58,6 +58,7 @@ struct wm8978_priv { unsigned int f_mclk; unsigned int f_256fs; unsigned int f_opclk; + int mclk_idx; enum wm8978_sysclk_src sysclk; u16 reg_cache[WM8978_CACHEREGNUM]; }; @@ -402,6 +403,35 @@ static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, pll_div->k = k; } + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * find index >= idx, such that, for a given f_out, + * 3 * f_mclk / 4 <= f_PLLOUT < 13 * f_mclk / 4 + * f_out can be f_256fs or f_opclk, currently only used for f_256fs. Can be + * generalised for f_opclk with suitable coefficient arrays, but currently + * the OPCLK divisor is calculated directly, not iteratively. + */ +static int wm8978_enum_mclk(unsigned int f_out, unsigned int f_mclk, + unsigned int *f_pllout) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + unsigned int f_pllout_x4 = 4 * f_out * mclk_numerator[i] / + mclk_denominator[i]; + if (3 * f_mclk <= f_pllout_x4 && f_pllout_x4 < 13 * f_mclk) { + *f_pllout = f_pllout_x4 / 4; + return i; + } + } + + return -EINVAL; +} + /* * Calculate internal frequencies and dividers, according to Figure 40 * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 @@ -412,12 +442,16 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) struct wm8978_pll_div pll_div; unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk, f_256fs = wm8978->f_256fs; - unsigned int f2, opclk_div; + unsigned int f2; if (!f_mclk) return -EINVAL; if (f_opclk) { + unsigned int opclk_div; + /* Cannot set up MCLK divider now, do later */ + wm8978->mclk_idx = -1; + /* * The user needs OPCLK. Choose OPCLKDIV to put * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. @@ -444,7 +478,7 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) wm8978->f_pllout = f_opclk * opclk_div; } else if (f_256fs) { /* - * Not using OPCLK, choose R: + * Not using OPCLK, but PLL is used for the codec, choose R: * 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12. * f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where * prescale = 1, or prescale = 2. Prescale is calculated inside @@ -453,18 +487,11 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) * f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK * must be 3.781MHz <= f_MCLK <= 32.768MHz */ - if (48 * f_256fs < 3 * f_mclk || 4 * f_256fs >= 13 * f_mclk) - return -EINVAL; + int idx = wm8978_enum_mclk(f_256fs, f_mclk, &wm8978->f_pllout); + if (idx < 0) + return idx; - /* - * MCLKDIV will be selected in .hw_params(), just choose a - * suitable f_PLLOUT - */ - if (4 * f_256fs < 3 * f_mclk) - /* Will have to use MCLKDIV */ - wm8978->f_pllout = wm8978->f_mclk * 3 / 4; - else - wm8978->f_pllout = f_256fs; + wm8978->mclk_idx = idx; /* GPIO1 into default mode as input - before configuring PLL */ snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); @@ -515,6 +542,20 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, wm8978->f_opclk = div; if (wm8978->f_mclk) + /* + * We know the MCLK frequency, the user has requested + * OPCLK, configure the PLL based on that and start it + * and OPCLK immediately. We will configure PLL to match + * user-requested OPCLK frquency as good as possible. + * In fact, it is likely, that matching the sampling + * rate, when it becomes known, is more important, and + * we will not be reconfiguring PLL then, because we + * must not interrupt OPCLK. But it should be fine, + * because typically the user will request OPCLK to run + * at 256fs or 512fs, and for these cases we will also + * find an exact MCLK divider configuration - it will + * be equal to or double the OPCLK divisor. + */ ret = wm8978_configure_pll(codec); break; case WM8978_BCLKDIV: @@ -640,10 +681,6 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -/* MCLK dividers */ -static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; -static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; - /* * Set PCM DAI bit size and sample rate. */ @@ -709,9 +746,11 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, wm8978->f_256fs = params_rate(params) * 256; if (wm8978->sysclk == WM8978_MCLK) { + wm8978->mclk_idx = -1; f_sel = wm8978->f_mclk; } else { if (!wm8978->f_pllout) { + /* We only enter here, if OPCLK is not used */ int ret = wm8978_configure_pll(codec); if (ret < 0) return ret; @@ -719,32 +758,34 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, f_sel = wm8978->f_pllout; } - /* - * In some cases it is possible to reconfigure PLL to a higher frequency - * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or - * 512 * fs, so, we should be fine. - */ - if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) - return -EINVAL; + if (wm8978->mclk_idx < 0) { + /* Either MCLK is used directly, or OPCLK is used */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; - for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { - diff = abs(wm8978->f_256fs * 3 - - f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); - if (diff < diff_best) { - diff_best = diff; - best = i; - } + if (diff < diff_best) { + diff_best = diff; + best = i; + } - if (!diff) - break; + if (!diff) + break; + } + } else { + /* OPCLK not used, codec driven by PLL */ + best = wm8978->mclk_idx; + diff = 0; } if (diff) - dev_warn(codec->dev, "Imprecise clock: %u%s\n", - f_sel * mclk_denominator[best] / mclk_numerator[best], - wm8978->sysclk == WM8978_MCLK ? - ", consider using PLL" : ""); + dev_warn(codec->dev, "Imprecise sampling rate: %uHz%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best] / 256, + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, params_format(params), params_rate(params), best); -- cgit v0.10.2 From 2f1ff6614cb5938e5c5760358752d92deb67fb63 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sun, 31 Jan 2010 12:02:12 -0800 Subject: ASoC: Fix continuation line formats String constants that are continued on subsequent lines with \ are not good. Signed-off-by: Joe Perches Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index cf0dfb7..67cbfe7 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -349,9 +349,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \ size, &sport_handle->tx_dma_phy, GFP_KERNEL); if (!sport_handle->tx_dma_buf) { - pr_err("Failed to allocate memory for tx dma \ - buf - Please increase uncached DMA \ - memory region\n"); + pr_err("Failed to allocate memory for tx dma buf - Please increase uncached DMA memory region\n"); return -ENOMEM; } else memset(sport_handle->tx_dma_buf, 0, size); @@ -362,9 +360,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) sport_handle->rx_dma_buf = dma_alloc_coherent(NULL, \ size, &sport_handle->rx_dma_phy, GFP_KERNEL); if (!sport_handle->rx_dma_buf) { - pr_err("Failed to allocate memory for rx dma \ - buf - Please increase uncached DMA \ - memory region\n"); + pr_err("Failed to allocate memory for rx dma buf - Please increase uncached DMA memory region\n"); return -ENOMEM; } else memset(sport_handle->rx_dma_buf, 0, size); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 62fbb84..c6c6a4a7 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -207,8 +207,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) { - pr_err("Failed to allocate dma memory \ - Please increase uncached DMA memory region\n"); + pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n"); return -ENOMEM; } buf->bytes = size; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index a8c73cb..5e03bb2 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -244,8 +244,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_coherent(pcm->card->dev, size * 4, &buf->addr, GFP_KERNEL); if (!buf->area) { - pr_err("Failed to allocate dma memory \ - Please increase uncached DMA memory region\n"); + pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n"); return -ENOMEM; } buf->bytes = size; -- cgit v0.10.2 From 3ed7074c4cc0de5ba77e180e5d96c23ef96859f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 20 Jan 2010 17:39:45 +0000 Subject: ASoC: Improved wm_hubs headphone handling Perform DC servo offset calibration using a series update sequence rather than startup update sequence, tuning the configuration of the WM8993 DC servo to make best use of this. Also introduce currently unused data allowing us to correct for any systematic errors in the DC servo calibration results and an alternative startup path for the headphone output which performs better with some chip revisions. The alternative setup sequence is enabled for WM8993. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 828d817..bacfc2f 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -213,6 +213,7 @@ static struct { }; struct wm8993_priv { + struct wm_hubs_data hubs_data; u16 reg_cache[WM8993_REGISTER_COUNT]; struct wm8993_platform_data pdata; struct snd_soc_codec codec; @@ -997,6 +998,11 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Tune DC servo configuration */ + snd_soc_write(codec, 0x44, 3); + snd_soc_write(codec, 0x56, 3); + snd_soc_write(codec, 0x44, 0); + /* Bring up VMID with fast soft start */ snd_soc_update_bits(codec, WM8993_ANTIPOP2, WM8993_STARTUP_BIAS_ENA | @@ -1591,6 +1597,8 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, codec->num_dai = 1; codec->private_data = wm8993; + wm8993->hubs_data.hp_startup_mode = 1; + memcpy(wm8993->reg_cache, wm8993_reg_defaults, sizeof(wm8993->reg_cache)); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index a67319d..0ad9f5d 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -68,24 +68,77 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) int count = 0; dev_dbg(codec->dev, "Waiting for DC servo...\n"); + do { count++; msleep(1); reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); - dev_dbg(codec->dev, "DC servo status: %x\n", reg); - } while ((reg & WM8993_DCS_CAL_COMPLETE_MASK) - != WM8993_DCS_CAL_COMPLETE_MASK && count < 1000); + dev_dbg(codec->dev, "DC servo: %x\n", reg); + } while (reg & WM8993_DCS_DATAPATH_BUSY); - if ((reg & WM8993_DCS_CAL_COMPLETE_MASK) - != WM8993_DCS_CAL_COMPLETE_MASK) + if (reg & WM8993_DCS_DATAPATH_BUSY) dev_err(codec->dev, "Timed out waiting for DC Servo\n"); } /* + * Startup calibration of the DC servo + */ +static void calibrate_dc_servo(struct snd_soc_codec *codec) +{ + struct wm_hubs_data *hubs = codec->private_data; + u16 reg, dcs_cfg; + + /* Set for 32 series updates */ + snd_soc_update_bits(codec, WM8993_DC_SERVO_1, + WM8993_DCS_SERIES_NO_01_MASK, + 32 << WM8993_DCS_SERIES_NO_01_SHIFT); + + /* Enable the DC servo. Write all bits to avoid triggering startup + * or write calibration. + */ + snd_soc_update_bits(codec, WM8993_DC_SERVO_0, + 0xFFFF, + WM8993_DCS_ENA_CHAN_0 | + WM8993_DCS_ENA_CHAN_1 | + WM8993_DCS_TRIG_SERIES_1 | + WM8993_DCS_TRIG_SERIES_0); + + wait_for_dc_servo(codec); + + /* Apply correction to DC servo result */ + if (hubs->dcs_codes) { + dev_dbg(codec->dev, "Applying %d code DC servo correction\n", + hubs->dcs_codes); + + /* HPOUT1L */ + reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) & + WM8993_DCS_INTEG_CHAN_0_MASK;; + reg += hubs->dcs_codes; + dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + + /* HPOUT1R */ + reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & + WM8993_DCS_INTEG_CHAN_1_MASK; + reg += hubs->dcs_codes; + dcs_cfg |= reg; + + /* Do it */ + snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); + snd_soc_update_bits(codec, WM8993_DC_SERVO_0, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1); + + wait_for_dc_servo(codec); + } +} + +/* * Update the DC servo calibration on gain changes */ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int ret; @@ -251,6 +304,47 @@ SOC_SINGLE_TLV("LINEOUT2 Volume", WM8993_LINE_OUTPUTS_VOLUME, 0, 1, 1, line_tlv), }; +static int hp_supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = codec->private_data; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (hubs->hp_startup_mode) { + case 0: + break; + case 1: + /* Enable the headphone amp */ + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, + WM8993_HPOUT1L_ENA | + WM8993_HPOUT1R_ENA, + WM8993_HPOUT1L_ENA | + WM8993_HPOUT1R_ENA); + + /* Enable the second stage */ + snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0, + WM8993_HPOUT1L_DLY | + WM8993_HPOUT1R_DLY, + WM8993_HPOUT1L_DLY | + WM8993_HPOUT1R_DLY); + break; + default: + dev_err(codec->dev, "Unknown HP startup mode %d\n", + hubs->hp_startup_mode); + break; + } + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, + WM8993_CP_ENA, 0); + break; + } + + return 0; +} + static int hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -271,14 +365,11 @@ static int hp_event(struct snd_soc_dapm_widget *w, reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY; snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); - /* Start the DC servo */ - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - 0xFFFF, - WM8993_DCS_ENA_CHAN_0 | - WM8993_DCS_ENA_CHAN_1 | - WM8993_DCS_TRIG_STARTUP_1 | - WM8993_DCS_TRIG_STARTUP_0); - wait_for_dc_servo(codec); + /* Smallest supported update interval */ + snd_soc_update_bits(codec, WM8993_DC_SERVO_1, + WM8993_DCS_TIMER_PERIOD_01_MASK, 1); + + calibrate_dc_servo(codec); reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT | WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT; @@ -286,23 +377,19 @@ static int hp_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: - reg &= ~(WM8993_HPOUT1L_RMV_SHORT | - WM8993_HPOUT1L_DLY | - WM8993_HPOUT1L_OUTP | - WM8993_HPOUT1R_RMV_SHORT | - WM8993_HPOUT1R_DLY | - WM8993_HPOUT1R_OUTP); + snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0, + WM8993_HPOUT1L_DLY | + WM8993_HPOUT1R_DLY | + WM8993_HPOUT1L_RMV_SHORT | + WM8993_HPOUT1R_RMV_SHORT, 0); - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - 0xffff, 0); + snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0, + WM8993_HPOUT1L_OUTP | + WM8993_HPOUT1R_OUTP, 0); - snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA, 0); - - snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, - WM8993_CP_ENA, 0); break; } @@ -473,6 +560,8 @@ SND_SOC_DAPM_MIXER("Right Output Mixer", WM8993_POWER_MANAGEMENT_3, 4, 0, SND_SOC_DAPM_PGA("Left Output PGA", WM8993_POWER_MANAGEMENT_3, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Output PGA", WM8993_POWER_MANAGEMENT_3, 6, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Headphone Supply", SND_SOC_NOPM, 0, 0, hp_supply_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0, hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -626,6 +715,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, { "Headphone PGA", NULL, "CLK_SYS" }, + { "Headphone PGA", NULL, "Headphone Supply" }, { "HPOUT1L", NULL, "Headphone PGA" }, { "HPOUT1R", NULL, "Headphone PGA" }, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 36d3fba..420104f 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -18,6 +18,12 @@ struct snd_soc_codec; extern const unsigned int wm_hubs_spkmix_tlv[]; +/* This *must* be the first element of the codec->private_data struct */ +struct wm_hubs_data { + int dcs_codes; + int hp_startup_mode; +}; + extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int); extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, -- cgit v0.10.2 From be587ef4f20cb5a0e42264909fa702a24081a160 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Feb 2010 18:31:06 +0000 Subject: ASoC: Activate DCS correction for WM8993 Use a two code correction for optimal performance. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index bacfc2f..61239e0 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1,7 +1,7 @@ /* * wm8993.c -- WM8993 ALSA SoC audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009, 2010 Wolfson Microelectronics plc * * Author: Mark Brown * @@ -1598,6 +1598,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, codec->private_data = wm8993; wm8993->hubs_data.hp_startup_mode = 1; + wm8993->hubs_data.dcs_codes = -2; memcpy(wm8993->reg_cache, wm8993_reg_defaults, sizeof(wm8993->reg_cache)); -- cgit v0.10.2 From 9e6e96a197a03752d39a63e4f83e0b707ccedad7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Jan 2010 17:47:12 +0000 Subject: ASoC: Add WM8994 CODEC driver The WM8994 is a highly integrated ultra-low power hi-fi audio subsystem designed for smartphones and other portable devices rich in multimedia features. It provides advanced digital mixing facilities enabling low power high quality interconnection of CPU, baseband and other audio sources through flexible digital and analogue routing, and integrates a class W headphone driver and stereo class D speaker drivers. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0aad72f..6b8a101 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,6 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C + select SND_SOC_WM8994 if I2C select SND_SOC_WM9081 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS @@ -243,6 +244,9 @@ config SND_SOC_WM8990 config SND_SOC_WM8993 tristate +config SND_SOC_WM8994 + tristate + config SND_SOC_WM9081 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fbd290e..209dd6c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-wm8978-objs := wm8978.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8993-objs := wm8993.o +snd-soc-wm8994-objs := wm8994.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o @@ -108,6 +109,7 @@ obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o +obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c new file mode 100644 index 0000000..5dd4b29 --- /dev/null +++ b/sound/soc/codecs/wm8994.c @@ -0,0 +1,3870 @@ +/* + * wm8994.c -- WM8994 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "wm8994.h" +#include "wm_hubs.h" + +static struct snd_soc_codec *wm8994_codec; +struct snd_soc_codec_device soc_codec_dev_wm8994; + +struct fll_config { + int src; + int in; + int out; +}; + +#define WM8994_NUM_DRC 3 +#define WM8994_NUM_EQ 3 + +static int wm8994_drc_base[] = { + WM8994_AIF1_DRC1_1, + WM8994_AIF1_DRC2_1, + WM8994_AIF2_DRC_1, +}; + +static int wm8994_retune_mobile_base[] = { + WM8994_AIF1_DAC1_EQ_GAINS_1, + WM8994_AIF1_DAC2_EQ_GAINS_1, + WM8994_AIF2_EQ_GAINS_1, +}; + +#define WM8994_REG_CACHE_SIZE 0x621 + +/* codec private data */ +struct wm8994_priv { + struct wm_hubs_data hubs; + struct snd_soc_codec codec; + u16 reg_cache[WM8994_REG_CACHE_SIZE + 1]; + int sysclk[2]; + int sysclk_rate[2]; + int mclk[2]; + int aifclk[2]; + struct fll_config fll[2], fll_suspend[2]; + + int dac_rates[2]; + int lrclk_shared[2]; + + /* Platform dependant DRC configuration */ + const char **drc_texts; + int drc_cfg[WM8994_NUM_DRC]; + struct soc_enum drc_enum; + + /* Platform dependant ReTune mobile configuration */ + int num_retune_mobile_texts; + const char **retune_mobile_texts; + int retune_mobile_cfg[WM8994_NUM_EQ]; + struct soc_enum retune_mobile_enum; + + struct wm8994_pdata *pdata; +}; + +static struct { + unsigned short readable; /* Mask of readable bits */ + unsigned short writable; /* Mask of writable bits */ + unsigned short vol; /* Mask of volatile bits */ +} access_masks[] = { + { 0xFFFF, 0xFFFF, 0x0000 }, /* R0 - Software Reset */ + { 0x3B37, 0x3B37, 0x0000 }, /* R1 - Power Management (1) */ + { 0x6BF0, 0x6BF0, 0x0000 }, /* R2 - Power Management (2) */ + { 0x3FF0, 0x3FF0, 0x0000 }, /* R3 - Power Management (3) */ + { 0x3F3F, 0x3F3F, 0x0000 }, /* R4 - Power Management (4) */ + { 0x3F0F, 0x3F0F, 0x0000 }, /* R5 - Power Management (5) */ + { 0x003F, 0x003F, 0x0000 }, /* R6 - Power Management (6) */ + { 0x0000, 0x0000, 0x0000 }, /* R7 */ + { 0x0000, 0x0000, 0x0000 }, /* R8 */ + { 0x0000, 0x0000, 0x0000 }, /* R9 */ + { 0x0000, 0x0000, 0x0000 }, /* R10 */ + { 0x0000, 0x0000, 0x0000 }, /* R11 */ + { 0x0000, 0x0000, 0x0000 }, /* R12 */ + { 0x0000, 0x0000, 0x0000 }, /* R13 */ + { 0x0000, 0x0000, 0x0000 }, /* R14 */ + { 0x0000, 0x0000, 0x0000 }, /* R15 */ + { 0x0000, 0x0000, 0x0000 }, /* R16 */ + { 0x0000, 0x0000, 0x0000 }, /* R17 */ + { 0x0000, 0x0000, 0x0000 }, /* R18 */ + { 0x0000, 0x0000, 0x0000 }, /* R19 */ + { 0x0000, 0x0000, 0x0000 }, /* R20 */ + { 0x01C0, 0x01C0, 0x0000 }, /* R21 - Input Mixer (1) */ + { 0x0000, 0x0000, 0x0000 }, /* R22 */ + { 0x0000, 0x0000, 0x0000 }, /* R23 */ + { 0x00DF, 0x01DF, 0x0000 }, /* R24 - Left Line Input 1&2 Volume */ + { 0x00DF, 0x01DF, 0x0000 }, /* R25 - Left Line Input 3&4 Volume */ + { 0x00DF, 0x01DF, 0x0000 }, /* R26 - Right Line Input 1&2 Volume */ + { 0x00DF, 0x01DF, 0x0000 }, /* R27 - Right Line Input 3&4 Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R28 - Left Output Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R29 - Right Output Volume */ + { 0x0077, 0x0077, 0x0000 }, /* R30 - Line Outputs Volume */ + { 0x0030, 0x0030, 0x0000 }, /* R31 - HPOUT2 Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R32 - Left OPGA Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R33 - Right OPGA Volume */ + { 0x007F, 0x007F, 0x0000 }, /* R34 - SPKMIXL Attenuation */ + { 0x017F, 0x017F, 0x0000 }, /* R35 - SPKMIXR Attenuation */ + { 0x003F, 0x003F, 0x0000 }, /* R36 - SPKOUT Mixers */ + { 0x003F, 0x003F, 0x0000 }, /* R37 - ClassD */ + { 0x00FF, 0x01FF, 0x0000 }, /* R38 - Speaker Volume Left */ + { 0x00FF, 0x01FF, 0x0000 }, /* R39 - Speaker Volume Right */ + { 0x00FF, 0x00FF, 0x0000 }, /* R40 - Input Mixer (2) */ + { 0x01B7, 0x01B7, 0x0000 }, /* R41 - Input Mixer (3) */ + { 0x01B7, 0x01B7, 0x0000 }, /* R42 - Input Mixer (4) */ + { 0x01C7, 0x01C7, 0x0000 }, /* R43 - Input Mixer (5) */ + { 0x01C7, 0x01C7, 0x0000 }, /* R44 - Input Mixer (6) */ + { 0x01FF, 0x01FF, 0x0000 }, /* R45 - Output Mixer (1) */ + { 0x01FF, 0x01FF, 0x0000 }, /* R46 - Output Mixer (2) */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R47 - Output Mixer (3) */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R48 - Output Mixer (4) */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R49 - Output Mixer (5) */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R50 - Output Mixer (6) */ + { 0x0038, 0x0038, 0x0000 }, /* R51 - HPOUT2 Mixer */ + { 0x0077, 0x0077, 0x0000 }, /* R52 - Line Mixer (1) */ + { 0x0077, 0x0077, 0x0000 }, /* R53 - Line Mixer (2) */ + { 0x03FF, 0x03FF, 0x0000 }, /* R54 - Speaker Mixer */ + { 0x00C1, 0x00C1, 0x0000 }, /* R55 - Additional Control */ + { 0x00F0, 0x00F0, 0x0000 }, /* R56 - AntiPOP (1) */ + { 0x01EF, 0x01EF, 0x0000 }, /* R57 - AntiPOP (2) */ + { 0x00FF, 0x00FF, 0x0000 }, /* R58 - MICBIAS */ + { 0x000F, 0x000F, 0x0000 }, /* R59 - LDO 1 */ + { 0x0007, 0x0007, 0x0000 }, /* R60 - LDO 2 */ + { 0x0000, 0x0000, 0x0000 }, /* R61 */ + { 0x0000, 0x0000, 0x0000 }, /* R62 */ + { 0x0000, 0x0000, 0x0000 }, /* R63 */ + { 0x0000, 0x0000, 0x0000 }, /* R64 */ + { 0x0000, 0x0000, 0x0000 }, /* R65 */ + { 0x0000, 0x0000, 0x0000 }, /* R66 */ + { 0x0000, 0x0000, 0x0000 }, /* R67 */ + { 0x0000, 0x0000, 0x0000 }, /* R68 */ + { 0x0000, 0x0000, 0x0000 }, /* R69 */ + { 0x0000, 0x0000, 0x0000 }, /* R70 */ + { 0x0000, 0x0000, 0x0000 }, /* R71 */ + { 0x0000, 0x0000, 0x0000 }, /* R72 */ + { 0x0000, 0x0000, 0x0000 }, /* R73 */ + { 0x0000, 0x0000, 0x0000 }, /* R74 */ + { 0x0000, 0x0000, 0x0000 }, /* R75 */ + { 0x8000, 0x8000, 0x0000 }, /* R76 - Charge Pump (1) */ + { 0x0000, 0x0000, 0x0000 }, /* R77 */ + { 0x0000, 0x0000, 0x0000 }, /* R78 */ + { 0x0000, 0x0000, 0x0000 }, /* R79 */ + { 0x0000, 0x0000, 0x0000 }, /* R80 */ + { 0x0301, 0x0301, 0x0000 }, /* R81 - Class W (1) */ + { 0x0000, 0x0000, 0x0000 }, /* R82 */ + { 0x0000, 0x0000, 0x0000 }, /* R83 */ + { 0x333F, 0x333F, 0x0000 }, /* R84 - DC Servo (1) */ + { 0x0FEF, 0x0FEF, 0x0000 }, /* R85 - DC Servo (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R86 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R87 - DC Servo (4) */ + { 0x0333, 0x0000, 0x0000 }, /* R88 - DC Servo Readback */ + { 0x0000, 0x0000, 0x0000 }, /* R89 */ + { 0x0000, 0x0000, 0x0000 }, /* R90 */ + { 0x0000, 0x0000, 0x0000 }, /* R91 */ + { 0x0000, 0x0000, 0x0000 }, /* R92 */ + { 0x0000, 0x0000, 0x0000 }, /* R93 */ + { 0x0000, 0x0000, 0x0000 }, /* R94 */ + { 0x0000, 0x0000, 0x0000 }, /* R95 */ + { 0x00EE, 0x00EE, 0x0000 }, /* R96 - Analogue HP (1) */ + { 0x0000, 0x0000, 0x0000 }, /* R97 */ + { 0x0000, 0x0000, 0x0000 }, /* R98 */ + { 0x0000, 0x0000, 0x0000 }, /* R99 */ + { 0x0000, 0x0000, 0x0000 }, /* R100 */ + { 0x0000, 0x0000, 0x0000 }, /* R101 */ + { 0x0000, 0x0000, 0x0000 }, /* R102 */ + { 0x0000, 0x0000, 0x0000 }, /* R103 */ + { 0x0000, 0x0000, 0x0000 }, /* R104 */ + { 0x0000, 0x0000, 0x0000 }, /* R105 */ + { 0x0000, 0x0000, 0x0000 }, /* R106 */ + { 0x0000, 0x0000, 0x0000 }, /* R107 */ + { 0x0000, 0x0000, 0x0000 }, /* R108 */ + { 0x0000, 0x0000, 0x0000 }, /* R109 */ + { 0x0000, 0x0000, 0x0000 }, /* R110 */ + { 0x0000, 0x0000, 0x0000 }, /* R111 */ + { 0x0000, 0x0000, 0x0000 }, /* R112 */ + { 0x0000, 0x0000, 0x0000 }, /* R113 */ + { 0x0000, 0x0000, 0x0000 }, /* R114 */ + { 0x0000, 0x0000, 0x0000 }, /* R115 */ + { 0x0000, 0x0000, 0x0000 }, /* R116 */ + { 0x0000, 0x0000, 0x0000 }, /* R117 */ + { 0x0000, 0x0000, 0x0000 }, /* R118 */ + { 0x0000, 0x0000, 0x0000 }, /* R119 */ + { 0x0000, 0x0000, 0x0000 }, /* R120 */ + { 0x0000, 0x0000, 0x0000 }, /* R121 */ + { 0x0000, 0x0000, 0x0000 }, /* R122 */ + { 0x0000, 0x0000, 0x0000 }, /* R123 */ + { 0x0000, 0x0000, 0x0000 }, /* R124 */ + { 0x0000, 0x0000, 0x0000 }, /* R125 */ + { 0x0000, 0x0000, 0x0000 }, /* R126 */ + { 0x0000, 0x0000, 0x0000 }, /* R127 */ + { 0x0000, 0x0000, 0x0000 }, /* R128 */ + { 0x0000, 0x0000, 0x0000 }, /* R129 */ + { 0x0000, 0x0000, 0x0000 }, /* R130 */ + { 0x0000, 0x0000, 0x0000 }, /* R131 */ + { 0x0000, 0x0000, 0x0000 }, /* R132 */ + { 0x0000, 0x0000, 0x0000 }, /* R133 */ + { 0x0000, 0x0000, 0x0000 }, /* R134 */ + { 0x0000, 0x0000, 0x0000 }, /* R135 */ + { 0x0000, 0x0000, 0x0000 }, /* R136 */ + { 0x0000, 0x0000, 0x0000 }, /* R137 */ + { 0x0000, 0x0000, 0x0000 }, /* R138 */ + { 0x0000, 0x0000, 0x0000 }, /* R139 */ + { 0x0000, 0x0000, 0x0000 }, /* R140 */ + { 0x0000, 0x0000, 0x0000 }, /* R141 */ + { 0x0000, 0x0000, 0x0000 }, /* R142 */ + { 0x0000, 0x0000, 0x0000 }, /* R143 */ + { 0x0000, 0x0000, 0x0000 }, /* R144 */ + { 0x0000, 0x0000, 0x0000 }, /* R145 */ + { 0x0000, 0x0000, 0x0000 }, /* R146 */ + { 0x0000, 0x0000, 0x0000 }, /* R147 */ + { 0x0000, 0x0000, 0x0000 }, /* R148 */ + { 0x0000, 0x0000, 0x0000 }, /* R149 */ + { 0x0000, 0x0000, 0x0000 }, /* R150 */ + { 0x0000, 0x0000, 0x0000 }, /* R151 */ + { 0x0000, 0x0000, 0x0000 }, /* R152 */ + { 0x0000, 0x0000, 0x0000 }, /* R153 */ + { 0x0000, 0x0000, 0x0000 }, /* R154 */ + { 0x0000, 0x0000, 0x0000 }, /* R155 */ + { 0x0000, 0x0000, 0x0000 }, /* R156 */ + { 0x0000, 0x0000, 0x0000 }, /* R157 */ + { 0x0000, 0x0000, 0x0000 }, /* R158 */ + { 0x0000, 0x0000, 0x0000 }, /* R159 */ + { 0x0000, 0x0000, 0x0000 }, /* R160 */ + { 0x0000, 0x0000, 0x0000 }, /* R161 */ + { 0x0000, 0x0000, 0x0000 }, /* R162 */ + { 0x0000, 0x0000, 0x0000 }, /* R163 */ + { 0x0000, 0x0000, 0x0000 }, /* R164 */ + { 0x0000, 0x0000, 0x0000 }, /* R165 */ + { 0x0000, 0x0000, 0x0000 }, /* R166 */ + { 0x0000, 0x0000, 0x0000 }, /* R167 */ + { 0x0000, 0x0000, 0x0000 }, /* R168 */ + { 0x0000, 0x0000, 0x0000 }, /* R169 */ + { 0x0000, 0x0000, 0x0000 }, /* R170 */ + { 0x0000, 0x0000, 0x0000 }, /* R171 */ + { 0x0000, 0x0000, 0x0000 }, /* R172 */ + { 0x0000, 0x0000, 0x0000 }, /* R173 */ + { 0x0000, 0x0000, 0x0000 }, /* R174 */ + { 0x0000, 0x0000, 0x0000 }, /* R175 */ + { 0x0000, 0x0000, 0x0000 }, /* R176 */ + { 0x0000, 0x0000, 0x0000 }, /* R177 */ + { 0x0000, 0x0000, 0x0000 }, /* R178 */ + { 0x0000, 0x0000, 0x0000 }, /* R179 */ + { 0x0000, 0x0000, 0x0000 }, /* R180 */ + { 0x0000, 0x0000, 0x0000 }, /* R181 */ + { 0x0000, 0x0000, 0x0000 }, /* R182 */ + { 0x0000, 0x0000, 0x0000 }, /* R183 */ + { 0x0000, 0x0000, 0x0000 }, /* R184 */ + { 0x0000, 0x0000, 0x0000 }, /* R185 */ + { 0x0000, 0x0000, 0x0000 }, /* R186 */ + { 0x0000, 0x0000, 0x0000 }, /* R187 */ + { 0x0000, 0x0000, 0x0000 }, /* R188 */ + { 0x0000, 0x0000, 0x0000 }, /* R189 */ + { 0x0000, 0x0000, 0x0000 }, /* R190 */ + { 0x0000, 0x0000, 0x0000 }, /* R191 */ + { 0x0000, 0x0000, 0x0000 }, /* R192 */ + { 0x0000, 0x0000, 0x0000 }, /* R193 */ + { 0x0000, 0x0000, 0x0000 }, /* R194 */ + { 0x0000, 0x0000, 0x0000 }, /* R195 */ + { 0x0000, 0x0000, 0x0000 }, /* R196 */ + { 0x0000, 0x0000, 0x0000 }, /* R197 */ + { 0x0000, 0x0000, 0x0000 }, /* R198 */ + { 0x0000, 0x0000, 0x0000 }, /* R199 */ + { 0x0000, 0x0000, 0x0000 }, /* R200 */ + { 0x0000, 0x0000, 0x0000 }, /* R201 */ + { 0x0000, 0x0000, 0x0000 }, /* R202 */ + { 0x0000, 0x0000, 0x0000 }, /* R203 */ + { 0x0000, 0x0000, 0x0000 }, /* R204 */ + { 0x0000, 0x0000, 0x0000 }, /* R205 */ + { 0x0000, 0x0000, 0x0000 }, /* R206 */ + { 0x0000, 0x0000, 0x0000 }, /* R207 */ + { 0x0000, 0x0000, 0x0000 }, /* R208 */ + { 0x0000, 0x0000, 0x0000 }, /* R209 */ + { 0x0000, 0x0000, 0x0000 }, /* R210 */ + { 0x0000, 0x0000, 0x0000 }, /* R211 */ + { 0x0000, 0x0000, 0x0000 }, /* R212 */ + { 0x0000, 0x0000, 0x0000 }, /* R213 */ + { 0x0000, 0x0000, 0x0000 }, /* R214 */ + { 0x0000, 0x0000, 0x0000 }, /* R215 */ + { 0x0000, 0x0000, 0x0000 }, /* R216 */ + { 0x0000, 0x0000, 0x0000 }, /* R217 */ + { 0x0000, 0x0000, 0x0000 }, /* R218 */ + { 0x0000, 0x0000, 0x0000 }, /* R219 */ + { 0x0000, 0x0000, 0x0000 }, /* R220 */ + { 0x0000, 0x0000, 0x0000 }, /* R221 */ + { 0x0000, 0x0000, 0x0000 }, /* R222 */ + { 0x0000, 0x0000, 0x0000 }, /* R223 */ + { 0x0000, 0x0000, 0x0000 }, /* R224 */ + { 0x0000, 0x0000, 0x0000 }, /* R225 */ + { 0x0000, 0x0000, 0x0000 }, /* R226 */ + { 0x0000, 0x0000, 0x0000 }, /* R227 */ + { 0x0000, 0x0000, 0x0000 }, /* R228 */ + { 0x0000, 0x0000, 0x0000 }, /* R229 */ + { 0x0000, 0x0000, 0x0000 }, /* R230 */ + { 0x0000, 0x0000, 0x0000 }, /* R231 */ + { 0x0000, 0x0000, 0x0000 }, /* R232 */ + { 0x0000, 0x0000, 0x0000 }, /* R233 */ + { 0x0000, 0x0000, 0x0000 }, /* R234 */ + { 0x0000, 0x0000, 0x0000 }, /* R235 */ + { 0x0000, 0x0000, 0x0000 }, /* R236 */ + { 0x0000, 0x0000, 0x0000 }, /* R237 */ + { 0x0000, 0x0000, 0x0000 }, /* R238 */ + { 0x0000, 0x0000, 0x0000 }, /* R239 */ + { 0x0000, 0x0000, 0x0000 }, /* R240 */ + { 0x0000, 0x0000, 0x0000 }, /* R241 */ + { 0x0000, 0x0000, 0x0000 }, /* R242 */ + { 0x0000, 0x0000, 0x0000 }, /* R243 */ + { 0x0000, 0x0000, 0x0000 }, /* R244 */ + { 0x0000, 0x0000, 0x0000 }, /* R245 */ + { 0x0000, 0x0000, 0x0000 }, /* R246 */ + { 0x0000, 0x0000, 0x0000 }, /* R247 */ + { 0x0000, 0x0000, 0x0000 }, /* R248 */ + { 0x0000, 0x0000, 0x0000 }, /* R249 */ + { 0x0000, 0x0000, 0x0000 }, /* R250 */ + { 0x0000, 0x0000, 0x0000 }, /* R251 */ + { 0x0000, 0x0000, 0x0000 }, /* R252 */ + { 0x0000, 0x0000, 0x0000 }, /* R253 */ + { 0x0000, 0x0000, 0x0000 }, /* R254 */ + { 0x0000, 0x0000, 0x0000 }, /* R255 */ + { 0x000F, 0x0000, 0x0000 }, /* R256 - Chip Revision */ + { 0x0074, 0x0074, 0x0000 }, /* R257 - Control Interface */ + { 0x0000, 0x0000, 0x0000 }, /* R258 */ + { 0x0000, 0x0000, 0x0000 }, /* R259 */ + { 0x0000, 0x0000, 0x0000 }, /* R260 */ + { 0x0000, 0x0000, 0x0000 }, /* R261 */ + { 0x0000, 0x0000, 0x0000 }, /* R262 */ + { 0x0000, 0x0000, 0x0000 }, /* R263 */ + { 0x0000, 0x0000, 0x0000 }, /* R264 */ + { 0x0000, 0x0000, 0x0000 }, /* R265 */ + { 0x0000, 0x0000, 0x0000 }, /* R266 */ + { 0x0000, 0x0000, 0x0000 }, /* R267 */ + { 0x0000, 0x0000, 0x0000 }, /* R268 */ + { 0x0000, 0x0000, 0x0000 }, /* R269 */ + { 0x0000, 0x0000, 0x0000 }, /* R270 */ + { 0x0000, 0x0000, 0x0000 }, /* R271 */ + { 0x807F, 0x837F, 0x0000 }, /* R272 - Write Sequencer Ctrl (1) */ + { 0x017F, 0x0000, 0x0000 }, /* R273 - Write Sequencer Ctrl (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R274 */ + { 0x0000, 0x0000, 0x0000 }, /* R275 */ + { 0x0000, 0x0000, 0x0000 }, /* R276 */ + { 0x0000, 0x0000, 0x0000 }, /* R277 */ + { 0x0000, 0x0000, 0x0000 }, /* R278 */ + { 0x0000, 0x0000, 0x0000 }, /* R279 */ + { 0x0000, 0x0000, 0x0000 }, /* R280 */ + { 0x0000, 0x0000, 0x0000 }, /* R281 */ + { 0x0000, 0x0000, 0x0000 }, /* R282 */ + { 0x0000, 0x0000, 0x0000 }, /* R283 */ + { 0x0000, 0x0000, 0x0000 }, /* R284 */ + { 0x0000, 0x0000, 0x0000 }, /* R285 */ + { 0x0000, 0x0000, 0x0000 }, /* R286 */ + { 0x0000, 0x0000, 0x0000 }, /* R287 */ + { 0x0000, 0x0000, 0x0000 }, /* R288 */ + { 0x0000, 0x0000, 0x0000 }, /* R289 */ + { 0x0000, 0x0000, 0x0000 }, /* R290 */ + { 0x0000, 0x0000, 0x0000 }, /* R291 */ + { 0x0000, 0x0000, 0x0000 }, /* R292 */ + { 0x0000, 0x0000, 0x0000 }, /* R293 */ + { 0x0000, 0x0000, 0x0000 }, /* R294 */ + { 0x0000, 0x0000, 0x0000 }, /* R295 */ + { 0x0000, 0x0000, 0x0000 }, /* R296 */ + { 0x0000, 0x0000, 0x0000 }, /* R297 */ + { 0x0000, 0x0000, 0x0000 }, /* R298 */ + { 0x0000, 0x0000, 0x0000 }, /* R299 */ + { 0x0000, 0x0000, 0x0000 }, /* R300 */ + { 0x0000, 0x0000, 0x0000 }, /* R301 */ + { 0x0000, 0x0000, 0x0000 }, /* R302 */ + { 0x0000, 0x0000, 0x0000 }, /* R303 */ + { 0x0000, 0x0000, 0x0000 }, /* R304 */ + { 0x0000, 0x0000, 0x0000 }, /* R305 */ + { 0x0000, 0x0000, 0x0000 }, /* R306 */ + { 0x0000, 0x0000, 0x0000 }, /* R307 */ + { 0x0000, 0x0000, 0x0000 }, /* R308 */ + { 0x0000, 0x0000, 0x0000 }, /* R309 */ + { 0x0000, 0x0000, 0x0000 }, /* R310 */ + { 0x0000, 0x0000, 0x0000 }, /* R311 */ + { 0x0000, 0x0000, 0x0000 }, /* R312 */ + { 0x0000, 0x0000, 0x0000 }, /* R313 */ + { 0x0000, 0x0000, 0x0000 }, /* R314 */ + { 0x0000, 0x0000, 0x0000 }, /* R315 */ + { 0x0000, 0x0000, 0x0000 }, /* R316 */ + { 0x0000, 0x0000, 0x0000 }, /* R317 */ + { 0x0000, 0x0000, 0x0000 }, /* R318 */ + { 0x0000, 0x0000, 0x0000 }, /* R319 */ + { 0x0000, 0x0000, 0x0000 }, /* R320 */ + { 0x0000, 0x0000, 0x0000 }, /* R321 */ + { 0x0000, 0x0000, 0x0000 }, /* R322 */ + { 0x0000, 0x0000, 0x0000 }, /* R323 */ + { 0x0000, 0x0000, 0x0000 }, /* R324 */ + { 0x0000, 0x0000, 0x0000 }, /* R325 */ + { 0x0000, 0x0000, 0x0000 }, /* R326 */ + { 0x0000, 0x0000, 0x0000 }, /* R327 */ + { 0x0000, 0x0000, 0x0000 }, /* R328 */ + { 0x0000, 0x0000, 0x0000 }, /* R329 */ + { 0x0000, 0x0000, 0x0000 }, /* R330 */ + { 0x0000, 0x0000, 0x0000 }, /* R331 */ + { 0x0000, 0x0000, 0x0000 }, /* R332 */ + { 0x0000, 0x0000, 0x0000 }, /* R333 */ + { 0x0000, 0x0000, 0x0000 }, /* R334 */ + { 0x0000, 0x0000, 0x0000 }, /* R335 */ + { 0x0000, 0x0000, 0x0000 }, /* R336 */ + { 0x0000, 0x0000, 0x0000 }, /* R337 */ + { 0x0000, 0x0000, 0x0000 }, /* R338 */ + { 0x0000, 0x0000, 0x0000 }, /* R339 */ + { 0x0000, 0x0000, 0x0000 }, /* R340 */ + { 0x0000, 0x0000, 0x0000 }, /* R341 */ + { 0x0000, 0x0000, 0x0000 }, /* R342 */ + { 0x0000, 0x0000, 0x0000 }, /* R343 */ + { 0x0000, 0x0000, 0x0000 }, /* R344 */ + { 0x0000, 0x0000, 0x0000 }, /* R345 */ + { 0x0000, 0x0000, 0x0000 }, /* R346 */ + { 0x0000, 0x0000, 0x0000 }, /* R347 */ + { 0x0000, 0x0000, 0x0000 }, /* R348 */ + { 0x0000, 0x0000, 0x0000 }, /* R349 */ + { 0x0000, 0x0000, 0x0000 }, /* R350 */ + { 0x0000, 0x0000, 0x0000 }, /* R351 */ + { 0x0000, 0x0000, 0x0000 }, /* R352 */ + { 0x0000, 0x0000, 0x0000 }, /* R353 */ + { 0x0000, 0x0000, 0x0000 }, /* R354 */ + { 0x0000, 0x0000, 0x0000 }, /* R355 */ + { 0x0000, 0x0000, 0x0000 }, /* R356 */ + { 0x0000, 0x0000, 0x0000 }, /* R357 */ + { 0x0000, 0x0000, 0x0000 }, /* R358 */ + { 0x0000, 0x0000, 0x0000 }, /* R359 */ + { 0x0000, 0x0000, 0x0000 }, /* R360 */ + { 0x0000, 0x0000, 0x0000 }, /* R361 */ + { 0x0000, 0x0000, 0x0000 }, /* R362 */ + { 0x0000, 0x0000, 0x0000 }, /* R363 */ + { 0x0000, 0x0000, 0x0000 }, /* R364 */ + { 0x0000, 0x0000, 0x0000 }, /* R365 */ + { 0x0000, 0x0000, 0x0000 }, /* R366 */ + { 0x0000, 0x0000, 0x0000 }, /* R367 */ + { 0x0000, 0x0000, 0x0000 }, /* R368 */ + { 0x0000, 0x0000, 0x0000 }, /* R369 */ + { 0x0000, 0x0000, 0x0000 }, /* R370 */ + { 0x0000, 0x0000, 0x0000 }, /* R371 */ + { 0x0000, 0x0000, 0x0000 }, /* R372 */ + { 0x0000, 0x0000, 0x0000 }, /* R373 */ + { 0x0000, 0x0000, 0x0000 }, /* R374 */ + { 0x0000, 0x0000, 0x0000 }, /* R375 */ + { 0x0000, 0x0000, 0x0000 }, /* R376 */ + { 0x0000, 0x0000, 0x0000 }, /* R377 */ + { 0x0000, 0x0000, 0x0000 }, /* R378 */ + { 0x0000, 0x0000, 0x0000 }, /* R379 */ + { 0x0000, 0x0000, 0x0000 }, /* R380 */ + { 0x0000, 0x0000, 0x0000 }, /* R381 */ + { 0x0000, 0x0000, 0x0000 }, /* R382 */ + { 0x0000, 0x0000, 0x0000 }, /* R383 */ + { 0x0000, 0x0000, 0x0000 }, /* R384 */ + { 0x0000, 0x0000, 0x0000 }, /* R385 */ + { 0x0000, 0x0000, 0x0000 }, /* R386 */ + { 0x0000, 0x0000, 0x0000 }, /* R387 */ + { 0x0000, 0x0000, 0x0000 }, /* R388 */ + { 0x0000, 0x0000, 0x0000 }, /* R389 */ + { 0x0000, 0x0000, 0x0000 }, /* R390 */ + { 0x0000, 0x0000, 0x0000 }, /* R391 */ + { 0x0000, 0x0000, 0x0000 }, /* R392 */ + { 0x0000, 0x0000, 0x0000 }, /* R393 */ + { 0x0000, 0x0000, 0x0000 }, /* R394 */ + { 0x0000, 0x0000, 0x0000 }, /* R395 */ + { 0x0000, 0x0000, 0x0000 }, /* R396 */ + { 0x0000, 0x0000, 0x0000 }, /* R397 */ + { 0x0000, 0x0000, 0x0000 }, /* R398 */ + { 0x0000, 0x0000, 0x0000 }, /* R399 */ + { 0x0000, 0x0000, 0x0000 }, /* R400 */ + { 0x0000, 0x0000, 0x0000 }, /* R401 */ + { 0x0000, 0x0000, 0x0000 }, /* R402 */ + { 0x0000, 0x0000, 0x0000 }, /* R403 */ + { 0x0000, 0x0000, 0x0000 }, /* R404 */ + { 0x0000, 0x0000, 0x0000 }, /* R405 */ + { 0x0000, 0x0000, 0x0000 }, /* R406 */ + { 0x0000, 0x0000, 0x0000 }, /* R407 */ + { 0x0000, 0x0000, 0x0000 }, /* R408 */ + { 0x0000, 0x0000, 0x0000 }, /* R409 */ + { 0x0000, 0x0000, 0x0000 }, /* R410 */ + { 0x0000, 0x0000, 0x0000 }, /* R411 */ + { 0x0000, 0x0000, 0x0000 }, /* R412 */ + { 0x0000, 0x0000, 0x0000 }, /* R413 */ + { 0x0000, 0x0000, 0x0000 }, /* R414 */ + { 0x0000, 0x0000, 0x0000 }, /* R415 */ + { 0x0000, 0x0000, 0x0000 }, /* R416 */ + { 0x0000, 0x0000, 0x0000 }, /* R417 */ + { 0x0000, 0x0000, 0x0000 }, /* R418 */ + { 0x0000, 0x0000, 0x0000 }, /* R419 */ + { 0x0000, 0x0000, 0x0000 }, /* R420 */ + { 0x0000, 0x0000, 0x0000 }, /* R421 */ + { 0x0000, 0x0000, 0x0000 }, /* R422 */ + { 0x0000, 0x0000, 0x0000 }, /* R423 */ + { 0x0000, 0x0000, 0x0000 }, /* R424 */ + { 0x0000, 0x0000, 0x0000 }, /* R425 */ + { 0x0000, 0x0000, 0x0000 }, /* R426 */ + { 0x0000, 0x0000, 0x0000 }, /* R427 */ + { 0x0000, 0x0000, 0x0000 }, /* R428 */ + { 0x0000, 0x0000, 0x0000 }, /* R429 */ + { 0x0000, 0x0000, 0x0000 }, /* R430 */ + { 0x0000, 0x0000, 0x0000 }, /* R431 */ + { 0x0000, 0x0000, 0x0000 }, /* R432 */ + { 0x0000, 0x0000, 0x0000 }, /* R433 */ + { 0x0000, 0x0000, 0x0000 }, /* R434 */ + { 0x0000, 0x0000, 0x0000 }, /* R435 */ + { 0x0000, 0x0000, 0x0000 }, /* R436 */ + { 0x0000, 0x0000, 0x0000 }, /* R437 */ + { 0x0000, 0x0000, 0x0000 }, /* R438 */ + { 0x0000, 0x0000, 0x0000 }, /* R439 */ + { 0x0000, 0x0000, 0x0000 }, /* R440 */ + { 0x0000, 0x0000, 0x0000 }, /* R441 */ + { 0x0000, 0x0000, 0x0000 }, /* R442 */ + { 0x0000, 0x0000, 0x0000 }, /* R443 */ + { 0x0000, 0x0000, 0x0000 }, /* R444 */ + { 0x0000, 0x0000, 0x0000 }, /* R445 */ + { 0x0000, 0x0000, 0x0000 }, /* R446 */ + { 0x0000, 0x0000, 0x0000 }, /* R447 */ + { 0x0000, 0x0000, 0x0000 }, /* R448 */ + { 0x0000, 0x0000, 0x0000 }, /* R449 */ + { 0x0000, 0x0000, 0x0000 }, /* R450 */ + { 0x0000, 0x0000, 0x0000 }, /* R451 */ + { 0x0000, 0x0000, 0x0000 }, /* R452 */ + { 0x0000, 0x0000, 0x0000 }, /* R453 */ + { 0x0000, 0x0000, 0x0000 }, /* R454 */ + { 0x0000, 0x0000, 0x0000 }, /* R455 */ + { 0x0000, 0x0000, 0x0000 }, /* R456 */ + { 0x0000, 0x0000, 0x0000 }, /* R457 */ + { 0x0000, 0x0000, 0x0000 }, /* R458 */ + { 0x0000, 0x0000, 0x0000 }, /* R459 */ + { 0x0000, 0x0000, 0x0000 }, /* R460 */ + { 0x0000, 0x0000, 0x0000 }, /* R461 */ + { 0x0000, 0x0000, 0x0000 }, /* R462 */ + { 0x0000, 0x0000, 0x0000 }, /* R463 */ + { 0x0000, 0x0000, 0x0000 }, /* R464 */ + { 0x0000, 0x0000, 0x0000 }, /* R465 */ + { 0x0000, 0x0000, 0x0000 }, /* R466 */ + { 0x0000, 0x0000, 0x0000 }, /* R467 */ + { 0x0000, 0x0000, 0x0000 }, /* R468 */ + { 0x0000, 0x0000, 0x0000 }, /* R469 */ + { 0x0000, 0x0000, 0x0000 }, /* R470 */ + { 0x0000, 0x0000, 0x0000 }, /* R471 */ + { 0x0000, 0x0000, 0x0000 }, /* R472 */ + { 0x0000, 0x0000, 0x0000 }, /* R473 */ + { 0x0000, 0x0000, 0x0000 }, /* R474 */ + { 0x0000, 0x0000, 0x0000 }, /* R475 */ + { 0x0000, 0x0000, 0x0000 }, /* R476 */ + { 0x0000, 0x0000, 0x0000 }, /* R477 */ + { 0x0000, 0x0000, 0x0000 }, /* R478 */ + { 0x0000, 0x0000, 0x0000 }, /* R479 */ + { 0x0000, 0x0000, 0x0000 }, /* R480 */ + { 0x0000, 0x0000, 0x0000 }, /* R481 */ + { 0x0000, 0x0000, 0x0000 }, /* R482 */ + { 0x0000, 0x0000, 0x0000 }, /* R483 */ + { 0x0000, 0x0000, 0x0000 }, /* R484 */ + { 0x0000, 0x0000, 0x0000 }, /* R485 */ + { 0x0000, 0x0000, 0x0000 }, /* R486 */ + { 0x0000, 0x0000, 0x0000 }, /* R487 */ + { 0x0000, 0x0000, 0x0000 }, /* R488 */ + { 0x0000, 0x0000, 0x0000 }, /* R489 */ + { 0x0000, 0x0000, 0x0000 }, /* R490 */ + { 0x0000, 0x0000, 0x0000 }, /* R491 */ + { 0x0000, 0x0000, 0x0000 }, /* R492 */ + { 0x0000, 0x0000, 0x0000 }, /* R493 */ + { 0x0000, 0x0000, 0x0000 }, /* R494 */ + { 0x0000, 0x0000, 0x0000 }, /* R495 */ + { 0x0000, 0x0000, 0x0000 }, /* R496 */ + { 0x0000, 0x0000, 0x0000 }, /* R497 */ + { 0x0000, 0x0000, 0x0000 }, /* R498 */ + { 0x0000, 0x0000, 0x0000 }, /* R499 */ + { 0x0000, 0x0000, 0x0000 }, /* R500 */ + { 0x0000, 0x0000, 0x0000 }, /* R501 */ + { 0x0000, 0x0000, 0x0000 }, /* R502 */ + { 0x0000, 0x0000, 0x0000 }, /* R503 */ + { 0x0000, 0x0000, 0x0000 }, /* R504 */ + { 0x0000, 0x0000, 0x0000 }, /* R505 */ + { 0x0000, 0x0000, 0x0000 }, /* R506 */ + { 0x0000, 0x0000, 0x0000 }, /* R507 */ + { 0x0000, 0x0000, 0x0000 }, /* R508 */ + { 0x0000, 0x0000, 0x0000 }, /* R509 */ + { 0x0000, 0x0000, 0x0000 }, /* R510 */ + { 0x0000, 0x0000, 0x0000 }, /* R511 */ + { 0x001F, 0x001F, 0x0000 }, /* R512 - AIF1 Clocking (1) */ + { 0x003F, 0x003F, 0x0000 }, /* R513 - AIF1 Clocking (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R514 */ + { 0x0000, 0x0000, 0x0000 }, /* R515 */ + { 0x001F, 0x001F, 0x0000 }, /* R516 - AIF2 Clocking (1) */ + { 0x003F, 0x003F, 0x0000 }, /* R517 - AIF2 Clocking (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R518 */ + { 0x0000, 0x0000, 0x0000 }, /* R519 */ + { 0x001F, 0x001F, 0x0000 }, /* R520 - Clocking (1) */ + { 0x0777, 0x0777, 0x0000 }, /* R521 - Clocking (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R522 */ + { 0x0000, 0x0000, 0x0000 }, /* R523 */ + { 0x0000, 0x0000, 0x0000 }, /* R524 */ + { 0x0000, 0x0000, 0x0000 }, /* R525 */ + { 0x0000, 0x0000, 0x0000 }, /* R526 */ + { 0x0000, 0x0000, 0x0000 }, /* R527 */ + { 0x00FF, 0x00FF, 0x0000 }, /* R528 - AIF1 Rate */ + { 0x00FF, 0x00FF, 0x0000 }, /* R529 - AIF2 Rate */ + { 0x000F, 0x0000, 0x0000 }, /* R530 - Rate Status */ + { 0x0000, 0x0000, 0x0000 }, /* R531 */ + { 0x0000, 0x0000, 0x0000 }, /* R532 */ + { 0x0000, 0x0000, 0x0000 }, /* R533 */ + { 0x0000, 0x0000, 0x0000 }, /* R534 */ + { 0x0000, 0x0000, 0x0000 }, /* R535 */ + { 0x0000, 0x0000, 0x0000 }, /* R536 */ + { 0x0000, 0x0000, 0x0000 }, /* R537 */ + { 0x0000, 0x0000, 0x0000 }, /* R538 */ + { 0x0000, 0x0000, 0x0000 }, /* R539 */ + { 0x0000, 0x0000, 0x0000 }, /* R540 */ + { 0x0000, 0x0000, 0x0000 }, /* R541 */ + { 0x0000, 0x0000, 0x0000 }, /* R542 */ + { 0x0000, 0x0000, 0x0000 }, /* R543 */ + { 0x0007, 0x0007, 0x0000 }, /* R544 - FLL1 Control (1) */ + { 0x3F77, 0x3F77, 0x0000 }, /* R545 - FLL1 Control (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R546 - FLL1 Control (3) */ + { 0x7FEF, 0x7FEF, 0x0000 }, /* R547 - FLL1 Control (4) */ + { 0x1FDB, 0x1FDB, 0x0000 }, /* R548 - FLL1 Control (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R549 */ + { 0x0000, 0x0000, 0x0000 }, /* R550 */ + { 0x0000, 0x0000, 0x0000 }, /* R551 */ + { 0x0000, 0x0000, 0x0000 }, /* R552 */ + { 0x0000, 0x0000, 0x0000 }, /* R553 */ + { 0x0000, 0x0000, 0x0000 }, /* R554 */ + { 0x0000, 0x0000, 0x0000 }, /* R555 */ + { 0x0000, 0x0000, 0x0000 }, /* R556 */ + { 0x0000, 0x0000, 0x0000 }, /* R557 */ + { 0x0000, 0x0000, 0x0000 }, /* R558 */ + { 0x0000, 0x0000, 0x0000 }, /* R559 */ + { 0x0000, 0x0000, 0x0000 }, /* R560 */ + { 0x0000, 0x0000, 0x0000 }, /* R561 */ + { 0x0000, 0x0000, 0x0000 }, /* R562 */ + { 0x0000, 0x0000, 0x0000 }, /* R563 */ + { 0x0000, 0x0000, 0x0000 }, /* R564 */ + { 0x0000, 0x0000, 0x0000 }, /* R565 */ + { 0x0000, 0x0000, 0x0000 }, /* R566 */ + { 0x0000, 0x0000, 0x0000 }, /* R567 */ + { 0x0000, 0x0000, 0x0000 }, /* R568 */ + { 0x0000, 0x0000, 0x0000 }, /* R569 */ + { 0x0000, 0x0000, 0x0000 }, /* R570 */ + { 0x0000, 0x0000, 0x0000 }, /* R571 */ + { 0x0000, 0x0000, 0x0000 }, /* R572 */ + { 0x0000, 0x0000, 0x0000 }, /* R573 */ + { 0x0000, 0x0000, 0x0000 }, /* R574 */ + { 0x0000, 0x0000, 0x0000 }, /* R575 */ + { 0x0007, 0x0007, 0x0000 }, /* R576 - FLL2 Control (1) */ + { 0x3F77, 0x3F77, 0x0000 }, /* R577 - FLL2 Control (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R578 - FLL2 Control (3) */ + { 0x7FEF, 0x7FEF, 0x0000 }, /* R579 - FLL2 Control (4) */ + { 0x1FDB, 0x1FDB, 0x0000 }, /* R580 - FLL2 Control (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R581 */ + { 0x0000, 0x0000, 0x0000 }, /* R582 */ + { 0x0000, 0x0000, 0x0000 }, /* R583 */ + { 0x0000, 0x0000, 0x0000 }, /* R584 */ + { 0x0000, 0x0000, 0x0000 }, /* R585 */ + { 0x0000, 0x0000, 0x0000 }, /* R586 */ + { 0x0000, 0x0000, 0x0000 }, /* R587 */ + { 0x0000, 0x0000, 0x0000 }, /* R588 */ + { 0x0000, 0x0000, 0x0000 }, /* R589 */ + { 0x0000, 0x0000, 0x0000 }, /* R590 */ + { 0x0000, 0x0000, 0x0000 }, /* R591 */ + { 0x0000, 0x0000, 0x0000 }, /* R592 */ + { 0x0000, 0x0000, 0x0000 }, /* R593 */ + { 0x0000, 0x0000, 0x0000 }, /* R594 */ + { 0x0000, 0x0000, 0x0000 }, /* R595 */ + { 0x0000, 0x0000, 0x0000 }, /* R596 */ + { 0x0000, 0x0000, 0x0000 }, /* R597 */ + { 0x0000, 0x0000, 0x0000 }, /* R598 */ + { 0x0000, 0x0000, 0x0000 }, /* R599 */ + { 0x0000, 0x0000, 0x0000 }, /* R600 */ + { 0x0000, 0x0000, 0x0000 }, /* R601 */ + { 0x0000, 0x0000, 0x0000 }, /* R602 */ + { 0x0000, 0x0000, 0x0000 }, /* R603 */ + { 0x0000, 0x0000, 0x0000 }, /* R604 */ + { 0x0000, 0x0000, 0x0000 }, /* R605 */ + { 0x0000, 0x0000, 0x0000 }, /* R606 */ + { 0x0000, 0x0000, 0x0000 }, /* R607 */ + { 0x0000, 0x0000, 0x0000 }, /* R608 */ + { 0x0000, 0x0000, 0x0000 }, /* R609 */ + { 0x0000, 0x0000, 0x0000 }, /* R610 */ + { 0x0000, 0x0000, 0x0000 }, /* R611 */ + { 0x0000, 0x0000, 0x0000 }, /* R612 */ + { 0x0000, 0x0000, 0x0000 }, /* R613 */ + { 0x0000, 0x0000, 0x0000 }, /* R614 */ + { 0x0000, 0x0000, 0x0000 }, /* R615 */ + { 0x0000, 0x0000, 0x0000 }, /* R616 */ + { 0x0000, 0x0000, 0x0000 }, /* R617 */ + { 0x0000, 0x0000, 0x0000 }, /* R618 */ + { 0x0000, 0x0000, 0x0000 }, /* R619 */ + { 0x0000, 0x0000, 0x0000 }, /* R620 */ + { 0x0000, 0x0000, 0x0000 }, /* R621 */ + { 0x0000, 0x0000, 0x0000 }, /* R622 */ + { 0x0000, 0x0000, 0x0000 }, /* R623 */ + { 0x0000, 0x0000, 0x0000 }, /* R624 */ + { 0x0000, 0x0000, 0x0000 }, /* R625 */ + { 0x0000, 0x0000, 0x0000 }, /* R626 */ + { 0x0000, 0x0000, 0x0000 }, /* R627 */ + { 0x0000, 0x0000, 0x0000 }, /* R628 */ + { 0x0000, 0x0000, 0x0000 }, /* R629 */ + { 0x0000, 0x0000, 0x0000 }, /* R630 */ + { 0x0000, 0x0000, 0x0000 }, /* R631 */ + { 0x0000, 0x0000, 0x0000 }, /* R632 */ + { 0x0000, 0x0000, 0x0000 }, /* R633 */ + { 0x0000, 0x0000, 0x0000 }, /* R634 */ + { 0x0000, 0x0000, 0x0000 }, /* R635 */ + { 0x0000, 0x0000, 0x0000 }, /* R636 */ + { 0x0000, 0x0000, 0x0000 }, /* R637 */ + { 0x0000, 0x0000, 0x0000 }, /* R638 */ + { 0x0000, 0x0000, 0x0000 }, /* R639 */ + { 0x0000, 0x0000, 0x0000 }, /* R640 */ + { 0x0000, 0x0000, 0x0000 }, /* R641 */ + { 0x0000, 0x0000, 0x0000 }, /* R642 */ + { 0x0000, 0x0000, 0x0000 }, /* R643 */ + { 0x0000, 0x0000, 0x0000 }, /* R644 */ + { 0x0000, 0x0000, 0x0000 }, /* R645 */ + { 0x0000, 0x0000, 0x0000 }, /* R646 */ + { 0x0000, 0x0000, 0x0000 }, /* R647 */ + { 0x0000, 0x0000, 0x0000 }, /* R648 */ + { 0x0000, 0x0000, 0x0000 }, /* R649 */ + { 0x0000, 0x0000, 0x0000 }, /* R650 */ + { 0x0000, 0x0000, 0x0000 }, /* R651 */ + { 0x0000, 0x0000, 0x0000 }, /* R652 */ + { 0x0000, 0x0000, 0x0000 }, /* R653 */ + { 0x0000, 0x0000, 0x0000 }, /* R654 */ + { 0x0000, 0x0000, 0x0000 }, /* R655 */ + { 0x0000, 0x0000, 0x0000 }, /* R656 */ + { 0x0000, 0x0000, 0x0000 }, /* R657 */ + { 0x0000, 0x0000, 0x0000 }, /* R658 */ + { 0x0000, 0x0000, 0x0000 }, /* R659 */ + { 0x0000, 0x0000, 0x0000 }, /* R660 */ + { 0x0000, 0x0000, 0x0000 }, /* R661 */ + { 0x0000, 0x0000, 0x0000 }, /* R662 */ + { 0x0000, 0x0000, 0x0000 }, /* R663 */ + { 0x0000, 0x0000, 0x0000 }, /* R664 */ + { 0x0000, 0x0000, 0x0000 }, /* R665 */ + { 0x0000, 0x0000, 0x0000 }, /* R666 */ + { 0x0000, 0x0000, 0x0000 }, /* R667 */ + { 0x0000, 0x0000, 0x0000 }, /* R668 */ + { 0x0000, 0x0000, 0x0000 }, /* R669 */ + { 0x0000, 0x0000, 0x0000 }, /* R670 */ + { 0x0000, 0x0000, 0x0000 }, /* R671 */ + { 0x0000, 0x0000, 0x0000 }, /* R672 */ + { 0x0000, 0x0000, 0x0000 }, /* R673 */ + { 0x0000, 0x0000, 0x0000 }, /* R674 */ + { 0x0000, 0x0000, 0x0000 }, /* R675 */ + { 0x0000, 0x0000, 0x0000 }, /* R676 */ + { 0x0000, 0x0000, 0x0000 }, /* R677 */ + { 0x0000, 0x0000, 0x0000 }, /* R678 */ + { 0x0000, 0x0000, 0x0000 }, /* R679 */ + { 0x0000, 0x0000, 0x0000 }, /* R680 */ + { 0x0000, 0x0000, 0x0000 }, /* R681 */ + { 0x0000, 0x0000, 0x0000 }, /* R682 */ + { 0x0000, 0x0000, 0x0000 }, /* R683 */ + { 0x0000, 0x0000, 0x0000 }, /* R684 */ + { 0x0000, 0x0000, 0x0000 }, /* R685 */ + { 0x0000, 0x0000, 0x0000 }, /* R686 */ + { 0x0000, 0x0000, 0x0000 }, /* R687 */ + { 0x0000, 0x0000, 0x0000 }, /* R688 */ + { 0x0000, 0x0000, 0x0000 }, /* R689 */ + { 0x0000, 0x0000, 0x0000 }, /* R690 */ + { 0x0000, 0x0000, 0x0000 }, /* R691 */ + { 0x0000, 0x0000, 0x0000 }, /* R692 */ + { 0x0000, 0x0000, 0x0000 }, /* R693 */ + { 0x0000, 0x0000, 0x0000 }, /* R694 */ + { 0x0000, 0x0000, 0x0000 }, /* R695 */ + { 0x0000, 0x0000, 0x0000 }, /* R696 */ + { 0x0000, 0x0000, 0x0000 }, /* R697 */ + { 0x0000, 0x0000, 0x0000 }, /* R698 */ + { 0x0000, 0x0000, 0x0000 }, /* R699 */ + { 0x0000, 0x0000, 0x0000 }, /* R700 */ + { 0x0000, 0x0000, 0x0000 }, /* R701 */ + { 0x0000, 0x0000, 0x0000 }, /* R702 */ + { 0x0000, 0x0000, 0x0000 }, /* R703 */ + { 0x0000, 0x0000, 0x0000 }, /* R704 */ + { 0x0000, 0x0000, 0x0000 }, /* R705 */ + { 0x0000, 0x0000, 0x0000 }, /* R706 */ + { 0x0000, 0x0000, 0x0000 }, /* R707 */ + { 0x0000, 0x0000, 0x0000 }, /* R708 */ + { 0x0000, 0x0000, 0x0000 }, /* R709 */ + { 0x0000, 0x0000, 0x0000 }, /* R710 */ + { 0x0000, 0x0000, 0x0000 }, /* R711 */ + { 0x0000, 0x0000, 0x0000 }, /* R712 */ + { 0x0000, 0x0000, 0x0000 }, /* R713 */ + { 0x0000, 0x0000, 0x0000 }, /* R714 */ + { 0x0000, 0x0000, 0x0000 }, /* R715 */ + { 0x0000, 0x0000, 0x0000 }, /* R716 */ + { 0x0000, 0x0000, 0x0000 }, /* R717 */ + { 0x0000, 0x0000, 0x0000 }, /* R718 */ + { 0x0000, 0x0000, 0x0000 }, /* R719 */ + { 0x0000, 0x0000, 0x0000 }, /* R720 */ + { 0x0000, 0x0000, 0x0000 }, /* R721 */ + { 0x0000, 0x0000, 0x0000 }, /* R722 */ + { 0x0000, 0x0000, 0x0000 }, /* R723 */ + { 0x0000, 0x0000, 0x0000 }, /* R724 */ + { 0x0000, 0x0000, 0x0000 }, /* R725 */ + { 0x0000, 0x0000, 0x0000 }, /* R726 */ + { 0x0000, 0x0000, 0x0000 }, /* R727 */ + { 0x0000, 0x0000, 0x0000 }, /* R728 */ + { 0x0000, 0x0000, 0x0000 }, /* R729 */ + { 0x0000, 0x0000, 0x0000 }, /* R730 */ + { 0x0000, 0x0000, 0x0000 }, /* R731 */ + { 0x0000, 0x0000, 0x0000 }, /* R732 */ + { 0x0000, 0x0000, 0x0000 }, /* R733 */ + { 0x0000, 0x0000, 0x0000 }, /* R734 */ + { 0x0000, 0x0000, 0x0000 }, /* R735 */ + { 0x0000, 0x0000, 0x0000 }, /* R736 */ + { 0x0000, 0x0000, 0x0000 }, /* R737 */ + { 0x0000, 0x0000, 0x0000 }, /* R738 */ + { 0x0000, 0x0000, 0x0000 }, /* R739 */ + { 0x0000, 0x0000, 0x0000 }, /* R740 */ + { 0x0000, 0x0000, 0x0000 }, /* R741 */ + { 0x0000, 0x0000, 0x0000 }, /* R742 */ + { 0x0000, 0x0000, 0x0000 }, /* R743 */ + { 0x0000, 0x0000, 0x0000 }, /* R744 */ + { 0x0000, 0x0000, 0x0000 }, /* R745 */ + { 0x0000, 0x0000, 0x0000 }, /* R746 */ + { 0x0000, 0x0000, 0x0000 }, /* R747 */ + { 0x0000, 0x0000, 0x0000 }, /* R748 */ + { 0x0000, 0x0000, 0x0000 }, /* R749 */ + { 0x0000, 0x0000, 0x0000 }, /* R750 */ + { 0x0000, 0x0000, 0x0000 }, /* R751 */ + { 0x0000, 0x0000, 0x0000 }, /* R752 */ + { 0x0000, 0x0000, 0x0000 }, /* R753 */ + { 0x0000, 0x0000, 0x0000 }, /* R754 */ + { 0x0000, 0x0000, 0x0000 }, /* R755 */ + { 0x0000, 0x0000, 0x0000 }, /* R756 */ + { 0x0000, 0x0000, 0x0000 }, /* R757 */ + { 0x0000, 0x0000, 0x0000 }, /* R758 */ + { 0x0000, 0x0000, 0x0000 }, /* R759 */ + { 0x0000, 0x0000, 0x0000 }, /* R760 */ + { 0x0000, 0x0000, 0x0000 }, /* R761 */ + { 0x0000, 0x0000, 0x0000 }, /* R762 */ + { 0x0000, 0x0000, 0x0000 }, /* R763 */ + { 0x0000, 0x0000, 0x0000 }, /* R764 */ + { 0x0000, 0x0000, 0x0000 }, /* R765 */ + { 0x0000, 0x0000, 0x0000 }, /* R766 */ + { 0x0000, 0x0000, 0x0000 }, /* R767 */ + { 0xE1F8, 0xE1F8, 0x0000 }, /* R768 - AIF1 Control (1) */ + { 0xCD1F, 0xCD1F, 0x0000 }, /* R769 - AIF1 Control (2) */ + { 0xF000, 0xF000, 0x0000 }, /* R770 - AIF1 Master/Slave */ + { 0x01F0, 0x01F0, 0x0000 }, /* R771 - AIF1 BCLK */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R772 - AIF1ADC LRCLK */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R773 - AIF1DAC LRCLK */ + { 0x0003, 0x0003, 0x0000 }, /* R774 - AIF1DAC Data */ + { 0x0003, 0x0003, 0x0000 }, /* R775 - AIF1ADC Data */ + { 0x0000, 0x0000, 0x0000 }, /* R776 */ + { 0x0000, 0x0000, 0x0000 }, /* R777 */ + { 0x0000, 0x0000, 0x0000 }, /* R778 */ + { 0x0000, 0x0000, 0x0000 }, /* R779 */ + { 0x0000, 0x0000, 0x0000 }, /* R780 */ + { 0x0000, 0x0000, 0x0000 }, /* R781 */ + { 0x0000, 0x0000, 0x0000 }, /* R782 */ + { 0x0000, 0x0000, 0x0000 }, /* R783 */ + { 0xF1F8, 0xF1F8, 0x0000 }, /* R784 - AIF2 Control (1) */ + { 0xFD1F, 0xFD1F, 0x0000 }, /* R785 - AIF2 Control (2) */ + { 0xF000, 0xF000, 0x0000 }, /* R786 - AIF2 Master/Slave */ + { 0x01F0, 0x01F0, 0x0000 }, /* R787 - AIF2 BCLK */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R788 - AIF2ADC LRCLK */ + { 0x0FFF, 0x0FFF, 0x0000 }, /* R789 - AIF2DAC LRCLK */ + { 0x0003, 0x0003, 0x0000 }, /* R790 - AIF2DAC Data */ + { 0x0003, 0x0003, 0x0000 }, /* R791 - AIF2ADC Data */ + { 0x0000, 0x0000, 0x0000 }, /* R792 */ + { 0x0000, 0x0000, 0x0000 }, /* R793 */ + { 0x0000, 0x0000, 0x0000 }, /* R794 */ + { 0x0000, 0x0000, 0x0000 }, /* R795 */ + { 0x0000, 0x0000, 0x0000 }, /* R796 */ + { 0x0000, 0x0000, 0x0000 }, /* R797 */ + { 0x0000, 0x0000, 0x0000 }, /* R798 */ + { 0x0000, 0x0000, 0x0000 }, /* R799 */ + { 0x0000, 0x0000, 0x0000 }, /* R800 */ + { 0x0000, 0x0000, 0x0000 }, /* R801 */ + { 0x0000, 0x0000, 0x0000 }, /* R802 */ + { 0x0000, 0x0000, 0x0000 }, /* R803 */ + { 0x0000, 0x0000, 0x0000 }, /* R804 */ + { 0x0000, 0x0000, 0x0000 }, /* R805 */ + { 0x0000, 0x0000, 0x0000 }, /* R806 */ + { 0x0000, 0x0000, 0x0000 }, /* R807 */ + { 0x0000, 0x0000, 0x0000 }, /* R808 */ + { 0x0000, 0x0000, 0x0000 }, /* R809 */ + { 0x0000, 0x0000, 0x0000 }, /* R810 */ + { 0x0000, 0x0000, 0x0000 }, /* R811 */ + { 0x0000, 0x0000, 0x0000 }, /* R812 */ + { 0x0000, 0x0000, 0x0000 }, /* R813 */ + { 0x0000, 0x0000, 0x0000 }, /* R814 */ + { 0x0000, 0x0000, 0x0000 }, /* R815 */ + { 0x0000, 0x0000, 0x0000 }, /* R816 */ + { 0x0000, 0x0000, 0x0000 }, /* R817 */ + { 0x0000, 0x0000, 0x0000 }, /* R818 */ + { 0x0000, 0x0000, 0x0000 }, /* R819 */ + { 0x0000, 0x0000, 0x0000 }, /* R820 */ + { 0x0000, 0x0000, 0x0000 }, /* R821 */ + { 0x0000, 0x0000, 0x0000 }, /* R822 */ + { 0x0000, 0x0000, 0x0000 }, /* R823 */ + { 0x0000, 0x0000, 0x0000 }, /* R824 */ + { 0x0000, 0x0000, 0x0000 }, /* R825 */ + { 0x0000, 0x0000, 0x0000 }, /* R826 */ + { 0x0000, 0x0000, 0x0000 }, /* R827 */ + { 0x0000, 0x0000, 0x0000 }, /* R828 */ + { 0x0000, 0x0000, 0x0000 }, /* R829 */ + { 0x0000, 0x0000, 0x0000 }, /* R830 */ + { 0x0000, 0x0000, 0x0000 }, /* R831 */ + { 0x0000, 0x0000, 0x0000 }, /* R832 */ + { 0x0000, 0x0000, 0x0000 }, /* R833 */ + { 0x0000, 0x0000, 0x0000 }, /* R834 */ + { 0x0000, 0x0000, 0x0000 }, /* R835 */ + { 0x0000, 0x0000, 0x0000 }, /* R836 */ + { 0x0000, 0x0000, 0x0000 }, /* R837 */ + { 0x0000, 0x0000, 0x0000 }, /* R838 */ + { 0x0000, 0x0000, 0x0000 }, /* R839 */ + { 0x0000, 0x0000, 0x0000 }, /* R840 */ + { 0x0000, 0x0000, 0x0000 }, /* R841 */ + { 0x0000, 0x0000, 0x0000 }, /* R842 */ + { 0x0000, 0x0000, 0x0000 }, /* R843 */ + { 0x0000, 0x0000, 0x0000 }, /* R844 */ + { 0x0000, 0x0000, 0x0000 }, /* R845 */ + { 0x0000, 0x0000, 0x0000 }, /* R846 */ + { 0x0000, 0x0000, 0x0000 }, /* R847 */ + { 0x0000, 0x0000, 0x0000 }, /* R848 */ + { 0x0000, 0x0000, 0x0000 }, /* R849 */ + { 0x0000, 0x0000, 0x0000 }, /* R850 */ + { 0x0000, 0x0000, 0x0000 }, /* R851 */ + { 0x0000, 0x0000, 0x0000 }, /* R852 */ + { 0x0000, 0x0000, 0x0000 }, /* R853 */ + { 0x0000, 0x0000, 0x0000 }, /* R854 */ + { 0x0000, 0x0000, 0x0000 }, /* R855 */ + { 0x0000, 0x0000, 0x0000 }, /* R856 */ + { 0x0000, 0x0000, 0x0000 }, /* R857 */ + { 0x0000, 0x0000, 0x0000 }, /* R858 */ + { 0x0000, 0x0000, 0x0000 }, /* R859 */ + { 0x0000, 0x0000, 0x0000 }, /* R860 */ + { 0x0000, 0x0000, 0x0000 }, /* R861 */ + { 0x0000, 0x0000, 0x0000 }, /* R862 */ + { 0x0000, 0x0000, 0x0000 }, /* R863 */ + { 0x0000, 0x0000, 0x0000 }, /* R864 */ + { 0x0000, 0x0000, 0x0000 }, /* R865 */ + { 0x0000, 0x0000, 0x0000 }, /* R866 */ + { 0x0000, 0x0000, 0x0000 }, /* R867 */ + { 0x0000, 0x0000, 0x0000 }, /* R868 */ + { 0x0000, 0x0000, 0x0000 }, /* R869 */ + { 0x0000, 0x0000, 0x0000 }, /* R870 */ + { 0x0000, 0x0000, 0x0000 }, /* R871 */ + { 0x0000, 0x0000, 0x0000 }, /* R872 */ + { 0x0000, 0x0000, 0x0000 }, /* R873 */ + { 0x0000, 0x0000, 0x0000 }, /* R874 */ + { 0x0000, 0x0000, 0x0000 }, /* R875 */ + { 0x0000, 0x0000, 0x0000 }, /* R876 */ + { 0x0000, 0x0000, 0x0000 }, /* R877 */ + { 0x0000, 0x0000, 0x0000 }, /* R878 */ + { 0x0000, 0x0000, 0x0000 }, /* R879 */ + { 0x0000, 0x0000, 0x0000 }, /* R880 */ + { 0x0000, 0x0000, 0x0000 }, /* R881 */ + { 0x0000, 0x0000, 0x0000 }, /* R882 */ + { 0x0000, 0x0000, 0x0000 }, /* R883 */ + { 0x0000, 0x0000, 0x0000 }, /* R884 */ + { 0x0000, 0x0000, 0x0000 }, /* R885 */ + { 0x0000, 0x0000, 0x0000 }, /* R886 */ + { 0x0000, 0x0000, 0x0000 }, /* R887 */ + { 0x0000, 0x0000, 0x0000 }, /* R888 */ + { 0x0000, 0x0000, 0x0000 }, /* R889 */ + { 0x0000, 0x0000, 0x0000 }, /* R890 */ + { 0x0000, 0x0000, 0x0000 }, /* R891 */ + { 0x0000, 0x0000, 0x0000 }, /* R892 */ + { 0x0000, 0x0000, 0x0000 }, /* R893 */ + { 0x0000, 0x0000, 0x0000 }, /* R894 */ + { 0x0000, 0x0000, 0x0000 }, /* R895 */ + { 0x0000, 0x0000, 0x0000 }, /* R896 */ + { 0x0000, 0x0000, 0x0000 }, /* R897 */ + { 0x0000, 0x0000, 0x0000 }, /* R898 */ + { 0x0000, 0x0000, 0x0000 }, /* R899 */ + { 0x0000, 0x0000, 0x0000 }, /* R900 */ + { 0x0000, 0x0000, 0x0000 }, /* R901 */ + { 0x0000, 0x0000, 0x0000 }, /* R902 */ + { 0x0000, 0x0000, 0x0000 }, /* R903 */ + { 0x0000, 0x0000, 0x0000 }, /* R904 */ + { 0x0000, 0x0000, 0x0000 }, /* R905 */ + { 0x0000, 0x0000, 0x0000 }, /* R906 */ + { 0x0000, 0x0000, 0x0000 }, /* R907 */ + { 0x0000, 0x0000, 0x0000 }, /* R908 */ + { 0x0000, 0x0000, 0x0000 }, /* R909 */ + { 0x0000, 0x0000, 0x0000 }, /* R910 */ + { 0x0000, 0x0000, 0x0000 }, /* R911 */ + { 0x0000, 0x0000, 0x0000 }, /* R912 */ + { 0x0000, 0x0000, 0x0000 }, /* R913 */ + { 0x0000, 0x0000, 0x0000 }, /* R914 */ + { 0x0000, 0x0000, 0x0000 }, /* R915 */ + { 0x0000, 0x0000, 0x0000 }, /* R916 */ + { 0x0000, 0x0000, 0x0000 }, /* R917 */ + { 0x0000, 0x0000, 0x0000 }, /* R918 */ + { 0x0000, 0x0000, 0x0000 }, /* R919 */ + { 0x0000, 0x0000, 0x0000 }, /* R920 */ + { 0x0000, 0x0000, 0x0000 }, /* R921 */ + { 0x0000, 0x0000, 0x0000 }, /* R922 */ + { 0x0000, 0x0000, 0x0000 }, /* R923 */ + { 0x0000, 0x0000, 0x0000 }, /* R924 */ + { 0x0000, 0x0000, 0x0000 }, /* R925 */ + { 0x0000, 0x0000, 0x0000 }, /* R926 */ + { 0x0000, 0x0000, 0x0000 }, /* R927 */ + { 0x0000, 0x0000, 0x0000 }, /* R928 */ + { 0x0000, 0x0000, 0x0000 }, /* R929 */ + { 0x0000, 0x0000, 0x0000 }, /* R930 */ + { 0x0000, 0x0000, 0x0000 }, /* R931 */ + { 0x0000, 0x0000, 0x0000 }, /* R932 */ + { 0x0000, 0x0000, 0x0000 }, /* R933 */ + { 0x0000, 0x0000, 0x0000 }, /* R934 */ + { 0x0000, 0x0000, 0x0000 }, /* R935 */ + { 0x0000, 0x0000, 0x0000 }, /* R936 */ + { 0x0000, 0x0000, 0x0000 }, /* R937 */ + { 0x0000, 0x0000, 0x0000 }, /* R938 */ + { 0x0000, 0x0000, 0x0000 }, /* R939 */ + { 0x0000, 0x0000, 0x0000 }, /* R940 */ + { 0x0000, 0x0000, 0x0000 }, /* R941 */ + { 0x0000, 0x0000, 0x0000 }, /* R942 */ + { 0x0000, 0x0000, 0x0000 }, /* R943 */ + { 0x0000, 0x0000, 0x0000 }, /* R944 */ + { 0x0000, 0x0000, 0x0000 }, /* R945 */ + { 0x0000, 0x0000, 0x0000 }, /* R946 */ + { 0x0000, 0x0000, 0x0000 }, /* R947 */ + { 0x0000, 0x0000, 0x0000 }, /* R948 */ + { 0x0000, 0x0000, 0x0000 }, /* R949 */ + { 0x0000, 0x0000, 0x0000 }, /* R950 */ + { 0x0000, 0x0000, 0x0000 }, /* R951 */ + { 0x0000, 0x0000, 0x0000 }, /* R952 */ + { 0x0000, 0x0000, 0x0000 }, /* R953 */ + { 0x0000, 0x0000, 0x0000 }, /* R954 */ + { 0x0000, 0x0000, 0x0000 }, /* R955 */ + { 0x0000, 0x0000, 0x0000 }, /* R956 */ + { 0x0000, 0x0000, 0x0000 }, /* R957 */ + { 0x0000, 0x0000, 0x0000 }, /* R958 */ + { 0x0000, 0x0000, 0x0000 }, /* R959 */ + { 0x0000, 0x0000, 0x0000 }, /* R960 */ + { 0x0000, 0x0000, 0x0000 }, /* R961 */ + { 0x0000, 0x0000, 0x0000 }, /* R962 */ + { 0x0000, 0x0000, 0x0000 }, /* R963 */ + { 0x0000, 0x0000, 0x0000 }, /* R964 */ + { 0x0000, 0x0000, 0x0000 }, /* R965 */ + { 0x0000, 0x0000, 0x0000 }, /* R966 */ + { 0x0000, 0x0000, 0x0000 }, /* R967 */ + { 0x0000, 0x0000, 0x0000 }, /* R968 */ + { 0x0000, 0x0000, 0x0000 }, /* R969 */ + { 0x0000, 0x0000, 0x0000 }, /* R970 */ + { 0x0000, 0x0000, 0x0000 }, /* R971 */ + { 0x0000, 0x0000, 0x0000 }, /* R972 */ + { 0x0000, 0x0000, 0x0000 }, /* R973 */ + { 0x0000, 0x0000, 0x0000 }, /* R974 */ + { 0x0000, 0x0000, 0x0000 }, /* R975 */ + { 0x0000, 0x0000, 0x0000 }, /* R976 */ + { 0x0000, 0x0000, 0x0000 }, /* R977 */ + { 0x0000, 0x0000, 0x0000 }, /* R978 */ + { 0x0000, 0x0000, 0x0000 }, /* R979 */ + { 0x0000, 0x0000, 0x0000 }, /* R980 */ + { 0x0000, 0x0000, 0x0000 }, /* R981 */ + { 0x0000, 0x0000, 0x0000 }, /* R982 */ + { 0x0000, 0x0000, 0x0000 }, /* R983 */ + { 0x0000, 0x0000, 0x0000 }, /* R984 */ + { 0x0000, 0x0000, 0x0000 }, /* R985 */ + { 0x0000, 0x0000, 0x0000 }, /* R986 */ + { 0x0000, 0x0000, 0x0000 }, /* R987 */ + { 0x0000, 0x0000, 0x0000 }, /* R988 */ + { 0x0000, 0x0000, 0x0000 }, /* R989 */ + { 0x0000, 0x0000, 0x0000 }, /* R990 */ + { 0x0000, 0x0000, 0x0000 }, /* R991 */ + { 0x0000, 0x0000, 0x0000 }, /* R992 */ + { 0x0000, 0x0000, 0x0000 }, /* R993 */ + { 0x0000, 0x0000, 0x0000 }, /* R994 */ + { 0x0000, 0x0000, 0x0000 }, /* R995 */ + { 0x0000, 0x0000, 0x0000 }, /* R996 */ + { 0x0000, 0x0000, 0x0000 }, /* R997 */ + { 0x0000, 0x0000, 0x0000 }, /* R998 */ + { 0x0000, 0x0000, 0x0000 }, /* R999 */ + { 0x0000, 0x0000, 0x0000 }, /* R1000 */ + { 0x0000, 0x0000, 0x0000 }, /* R1001 */ + { 0x0000, 0x0000, 0x0000 }, /* R1002 */ + { 0x0000, 0x0000, 0x0000 }, /* R1003 */ + { 0x0000, 0x0000, 0x0000 }, /* R1004 */ + { 0x0000, 0x0000, 0x0000 }, /* R1005 */ + { 0x0000, 0x0000, 0x0000 }, /* R1006 */ + { 0x0000, 0x0000, 0x0000 }, /* R1007 */ + { 0x0000, 0x0000, 0x0000 }, /* R1008 */ + { 0x0000, 0x0000, 0x0000 }, /* R1009 */ + { 0x0000, 0x0000, 0x0000 }, /* R1010 */ + { 0x0000, 0x0000, 0x0000 }, /* R1011 */ + { 0x0000, 0x0000, 0x0000 }, /* R1012 */ + { 0x0000, 0x0000, 0x0000 }, /* R1013 */ + { 0x0000, 0x0000, 0x0000 }, /* R1014 */ + { 0x0000, 0x0000, 0x0000 }, /* R1015 */ + { 0x0000, 0x0000, 0x0000 }, /* R1016 */ + { 0x0000, 0x0000, 0x0000 }, /* R1017 */ + { 0x0000, 0x0000, 0x0000 }, /* R1018 */ + { 0x0000, 0x0000, 0x0000 }, /* R1019 */ + { 0x0000, 0x0000, 0x0000 }, /* R1020 */ + { 0x0000, 0x0000, 0x0000 }, /* R1021 */ + { 0x0000, 0x0000, 0x0000 }, /* R1022 */ + { 0x0000, 0x0000, 0x0000 }, /* R1023 */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1024 - AIF1 ADC1 Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1025 - AIF1 ADC1 Right Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1026 - AIF1 DAC1 Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1027 - AIF1 DAC1 Right Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1028 - AIF1 ADC2 Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1029 - AIF1 ADC2 Right Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1030 - AIF1 DAC2 Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1031 - AIF1 DAC2 Right Volume */ + { 0x0000, 0x0000, 0x0000 }, /* R1032 */ + { 0x0000, 0x0000, 0x0000 }, /* R1033 */ + { 0x0000, 0x0000, 0x0000 }, /* R1034 */ + { 0x0000, 0x0000, 0x0000 }, /* R1035 */ + { 0x0000, 0x0000, 0x0000 }, /* R1036 */ + { 0x0000, 0x0000, 0x0000 }, /* R1037 */ + { 0x0000, 0x0000, 0x0000 }, /* R1038 */ + { 0x0000, 0x0000, 0x0000 }, /* R1039 */ + { 0xF800, 0xF800, 0x0000 }, /* R1040 - AIF1 ADC1 Filters */ + { 0x7800, 0x7800, 0x0000 }, /* R1041 - AIF1 ADC2 Filters */ + { 0x0000, 0x0000, 0x0000 }, /* R1042 */ + { 0x0000, 0x0000, 0x0000 }, /* R1043 */ + { 0x0000, 0x0000, 0x0000 }, /* R1044 */ + { 0x0000, 0x0000, 0x0000 }, /* R1045 */ + { 0x0000, 0x0000, 0x0000 }, /* R1046 */ + { 0x0000, 0x0000, 0x0000 }, /* R1047 */ + { 0x0000, 0x0000, 0x0000 }, /* R1048 */ + { 0x0000, 0x0000, 0x0000 }, /* R1049 */ + { 0x0000, 0x0000, 0x0000 }, /* R1050 */ + { 0x0000, 0x0000, 0x0000 }, /* R1051 */ + { 0x0000, 0x0000, 0x0000 }, /* R1052 */ + { 0x0000, 0x0000, 0x0000 }, /* R1053 */ + { 0x0000, 0x0000, 0x0000 }, /* R1054 */ + { 0x0000, 0x0000, 0x0000 }, /* R1055 */ + { 0x02B6, 0x02B6, 0x0000 }, /* R1056 - AIF1 DAC1 Filters (1) */ + { 0x3F00, 0x3F00, 0x0000 }, /* R1057 - AIF1 DAC1 Filters (2) */ + { 0x02B6, 0x02B6, 0x0000 }, /* R1058 - AIF1 DAC2 Filters (1) */ + { 0x3F00, 0x3F00, 0x0000 }, /* R1059 - AIF1 DAC2 Filters (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R1060 */ + { 0x0000, 0x0000, 0x0000 }, /* R1061 */ + { 0x0000, 0x0000, 0x0000 }, /* R1062 */ + { 0x0000, 0x0000, 0x0000 }, /* R1063 */ + { 0x0000, 0x0000, 0x0000 }, /* R1064 */ + { 0x0000, 0x0000, 0x0000 }, /* R1065 */ + { 0x0000, 0x0000, 0x0000 }, /* R1066 */ + { 0x0000, 0x0000, 0x0000 }, /* R1067 */ + { 0x0000, 0x0000, 0x0000 }, /* R1068 */ + { 0x0000, 0x0000, 0x0000 }, /* R1069 */ + { 0x0000, 0x0000, 0x0000 }, /* R1070 */ + { 0x0000, 0x0000, 0x0000 }, /* R1071 */ + { 0x0000, 0x0000, 0x0000 }, /* R1072 */ + { 0x0000, 0x0000, 0x0000 }, /* R1073 */ + { 0x0000, 0x0000, 0x0000 }, /* R1074 */ + { 0x0000, 0x0000, 0x0000 }, /* R1075 */ + { 0x0000, 0x0000, 0x0000 }, /* R1076 */ + { 0x0000, 0x0000, 0x0000 }, /* R1077 */ + { 0x0000, 0x0000, 0x0000 }, /* R1078 */ + { 0x0000, 0x0000, 0x0000 }, /* R1079 */ + { 0x0000, 0x0000, 0x0000 }, /* R1080 */ + { 0x0000, 0x0000, 0x0000 }, /* R1081 */ + { 0x0000, 0x0000, 0x0000 }, /* R1082 */ + { 0x0000, 0x0000, 0x0000 }, /* R1083 */ + { 0x0000, 0x0000, 0x0000 }, /* R1084 */ + { 0x0000, 0x0000, 0x0000 }, /* R1085 */ + { 0x0000, 0x0000, 0x0000 }, /* R1086 */ + { 0x0000, 0x0000, 0x0000 }, /* R1087 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1088 - AIF1 DRC1 (1) */ + { 0x1FFF, 0x1FFF, 0x0000 }, /* R1089 - AIF1 DRC1 (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1090 - AIF1 DRC1 (3) */ + { 0x07FF, 0x07FF, 0x0000 }, /* R1091 - AIF1 DRC1 (4) */ + { 0x03FF, 0x03FF, 0x0000 }, /* R1092 - AIF1 DRC1 (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R1093 */ + { 0x0000, 0x0000, 0x0000 }, /* R1094 */ + { 0x0000, 0x0000, 0x0000 }, /* R1095 */ + { 0x0000, 0x0000, 0x0000 }, /* R1096 */ + { 0x0000, 0x0000, 0x0000 }, /* R1097 */ + { 0x0000, 0x0000, 0x0000 }, /* R1098 */ + { 0x0000, 0x0000, 0x0000 }, /* R1099 */ + { 0x0000, 0x0000, 0x0000 }, /* R1100 */ + { 0x0000, 0x0000, 0x0000 }, /* R1101 */ + { 0x0000, 0x0000, 0x0000 }, /* R1102 */ + { 0x0000, 0x0000, 0x0000 }, /* R1103 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1104 - AIF1 DRC2 (1) */ + { 0x1FFF, 0x1FFF, 0x0000 }, /* R1105 - AIF1 DRC2 (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1106 - AIF1 DRC2 (3) */ + { 0x07FF, 0x07FF, 0x0000 }, /* R1107 - AIF1 DRC2 (4) */ + { 0x03FF, 0x03FF, 0x0000 }, /* R1108 - AIF1 DRC2 (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R1109 */ + { 0x0000, 0x0000, 0x0000 }, /* R1110 */ + { 0x0000, 0x0000, 0x0000 }, /* R1111 */ + { 0x0000, 0x0000, 0x0000 }, /* R1112 */ + { 0x0000, 0x0000, 0x0000 }, /* R1113 */ + { 0x0000, 0x0000, 0x0000 }, /* R1114 */ + { 0x0000, 0x0000, 0x0000 }, /* R1115 */ + { 0x0000, 0x0000, 0x0000 }, /* R1116 */ + { 0x0000, 0x0000, 0x0000 }, /* R1117 */ + { 0x0000, 0x0000, 0x0000 }, /* R1118 */ + { 0x0000, 0x0000, 0x0000 }, /* R1119 */ + { 0x0000, 0x0000, 0x0000 }, /* R1120 */ + { 0x0000, 0x0000, 0x0000 }, /* R1121 */ + { 0x0000, 0x0000, 0x0000 }, /* R1122 */ + { 0x0000, 0x0000, 0x0000 }, /* R1123 */ + { 0x0000, 0x0000, 0x0000 }, /* R1124 */ + { 0x0000, 0x0000, 0x0000 }, /* R1125 */ + { 0x0000, 0x0000, 0x0000 }, /* R1126 */ + { 0x0000, 0x0000, 0x0000 }, /* R1127 */ + { 0x0000, 0x0000, 0x0000 }, /* R1128 */ + { 0x0000, 0x0000, 0x0000 }, /* R1129 */ + { 0x0000, 0x0000, 0x0000 }, /* R1130 */ + { 0x0000, 0x0000, 0x0000 }, /* R1131 */ + { 0x0000, 0x0000, 0x0000 }, /* R1132 */ + { 0x0000, 0x0000, 0x0000 }, /* R1133 */ + { 0x0000, 0x0000, 0x0000 }, /* R1134 */ + { 0x0000, 0x0000, 0x0000 }, /* R1135 */ + { 0x0000, 0x0000, 0x0000 }, /* R1136 */ + { 0x0000, 0x0000, 0x0000 }, /* R1137 */ + { 0x0000, 0x0000, 0x0000 }, /* R1138 */ + { 0x0000, 0x0000, 0x0000 }, /* R1139 */ + { 0x0000, 0x0000, 0x0000 }, /* R1140 */ + { 0x0000, 0x0000, 0x0000 }, /* R1141 */ + { 0x0000, 0x0000, 0x0000 }, /* R1142 */ + { 0x0000, 0x0000, 0x0000 }, /* R1143 */ + { 0x0000, 0x0000, 0x0000 }, /* R1144 */ + { 0x0000, 0x0000, 0x0000 }, /* R1145 */ + { 0x0000, 0x0000, 0x0000 }, /* R1146 */ + { 0x0000, 0x0000, 0x0000 }, /* R1147 */ + { 0x0000, 0x0000, 0x0000 }, /* R1148 */ + { 0x0000, 0x0000, 0x0000 }, /* R1149 */ + { 0x0000, 0x0000, 0x0000 }, /* R1150 */ + { 0x0000, 0x0000, 0x0000 }, /* R1151 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1152 - AIF1 DAC1 EQ Gains (1) */ + { 0xFFC0, 0xFFC0, 0x0000 }, /* R1153 - AIF1 DAC1 EQ Gains (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1154 - AIF1 DAC1 EQ Band 1 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1155 - AIF1 DAC1 EQ Band 1 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1156 - AIF1 DAC1 EQ Band 1 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1157 - AIF1 DAC1 EQ Band 2 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1158 - AIF1 DAC1 EQ Band 2 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1159 - AIF1 DAC1 EQ Band 2 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1160 - AIF1 DAC1 EQ Band 2 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1161 - AIF1 DAC1 EQ Band 3 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1162 - AIF1 DAC1 EQ Band 3 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1163 - AIF1 DAC1 EQ Band 3 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1164 - AIF1 DAC1 EQ Band 3 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1165 - AIF1 DAC1 EQ Band 4 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1166 - AIF1 DAC1 EQ Band 4 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1167 - AIF1 DAC1 EQ Band 4 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1168 - AIF1 DAC1 EQ Band 4 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1169 - AIF1 DAC1 EQ Band 5 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1170 - AIF1 DAC1 EQ Band 5 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1171 - AIF1 DAC1 EQ Band 5 PG */ + { 0x0000, 0x0000, 0x0000 }, /* R1172 */ + { 0x0000, 0x0000, 0x0000 }, /* R1173 */ + { 0x0000, 0x0000, 0x0000 }, /* R1174 */ + { 0x0000, 0x0000, 0x0000 }, /* R1175 */ + { 0x0000, 0x0000, 0x0000 }, /* R1176 */ + { 0x0000, 0x0000, 0x0000 }, /* R1177 */ + { 0x0000, 0x0000, 0x0000 }, /* R1178 */ + { 0x0000, 0x0000, 0x0000 }, /* R1179 */ + { 0x0000, 0x0000, 0x0000 }, /* R1180 */ + { 0x0000, 0x0000, 0x0000 }, /* R1181 */ + { 0x0000, 0x0000, 0x0000 }, /* R1182 */ + { 0x0000, 0x0000, 0x0000 }, /* R1183 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1184 - AIF1 DAC2 EQ Gains (1) */ + { 0xFFC0, 0xFFC0, 0x0000 }, /* R1185 - AIF1 DAC2 EQ Gains (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1186 - AIF1 DAC2 EQ Band 1 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1187 - AIF1 DAC2 EQ Band 1 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1188 - AIF1 DAC2 EQ Band 1 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1189 - AIF1 DAC2 EQ Band 2 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1190 - AIF1 DAC2 EQ Band 2 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1191 - AIF1 DAC2 EQ Band 2 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1192 - AIF1 DAC2 EQ Band 2 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1193 - AIF1 DAC2 EQ Band 3 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1194 - AIF1 DAC2 EQ Band 3 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1195 - AIF1 DAC2 EQ Band 3 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1196 - AIF1 DAC2 EQ Band 3 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1197 - AIF1 DAC2 EQ Band 4 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1198 - AIF1 DAC2 EQ Band 4 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1199 - AIF1 DAC2 EQ Band 4 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1200 - AIF1 DAC2 EQ Band 4 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1201 - AIF1 DAC2 EQ Band 5 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1202 - AIF1 DAC2 EQ Band 5 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1203 - AIF1 DAC2 EQ Band 5 PG */ + { 0x0000, 0x0000, 0x0000 }, /* R1204 */ + { 0x0000, 0x0000, 0x0000 }, /* R1205 */ + { 0x0000, 0x0000, 0x0000 }, /* R1206 */ + { 0x0000, 0x0000, 0x0000 }, /* R1207 */ + { 0x0000, 0x0000, 0x0000 }, /* R1208 */ + { 0x0000, 0x0000, 0x0000 }, /* R1209 */ + { 0x0000, 0x0000, 0x0000 }, /* R1210 */ + { 0x0000, 0x0000, 0x0000 }, /* R1211 */ + { 0x0000, 0x0000, 0x0000 }, /* R1212 */ + { 0x0000, 0x0000, 0x0000 }, /* R1213 */ + { 0x0000, 0x0000, 0x0000 }, /* R1214 */ + { 0x0000, 0x0000, 0x0000 }, /* R1215 */ + { 0x0000, 0x0000, 0x0000 }, /* R1216 */ + { 0x0000, 0x0000, 0x0000 }, /* R1217 */ + { 0x0000, 0x0000, 0x0000 }, /* R1218 */ + { 0x0000, 0x0000, 0x0000 }, /* R1219 */ + { 0x0000, 0x0000, 0x0000 }, /* R1220 */ + { 0x0000, 0x0000, 0x0000 }, /* R1221 */ + { 0x0000, 0x0000, 0x0000 }, /* R1222 */ + { 0x0000, 0x0000, 0x0000 }, /* R1223 */ + { 0x0000, 0x0000, 0x0000 }, /* R1224 */ + { 0x0000, 0x0000, 0x0000 }, /* R1225 */ + { 0x0000, 0x0000, 0x0000 }, /* R1226 */ + { 0x0000, 0x0000, 0x0000 }, /* R1227 */ + { 0x0000, 0x0000, 0x0000 }, /* R1228 */ + { 0x0000, 0x0000, 0x0000 }, /* R1229 */ + { 0x0000, 0x0000, 0x0000 }, /* R1230 */ + { 0x0000, 0x0000, 0x0000 }, /* R1231 */ + { 0x0000, 0x0000, 0x0000 }, /* R1232 */ + { 0x0000, 0x0000, 0x0000 }, /* R1233 */ + { 0x0000, 0x0000, 0x0000 }, /* R1234 */ + { 0x0000, 0x0000, 0x0000 }, /* R1235 */ + { 0x0000, 0x0000, 0x0000 }, /* R1236 */ + { 0x0000, 0x0000, 0x0000 }, /* R1237 */ + { 0x0000, 0x0000, 0x0000 }, /* R1238 */ + { 0x0000, 0x0000, 0x0000 }, /* R1239 */ + { 0x0000, 0x0000, 0x0000 }, /* R1240 */ + { 0x0000, 0x0000, 0x0000 }, /* R1241 */ + { 0x0000, 0x0000, 0x0000 }, /* R1242 */ + { 0x0000, 0x0000, 0x0000 }, /* R1243 */ + { 0x0000, 0x0000, 0x0000 }, /* R1244 */ + { 0x0000, 0x0000, 0x0000 }, /* R1245 */ + { 0x0000, 0x0000, 0x0000 }, /* R1246 */ + { 0x0000, 0x0000, 0x0000 }, /* R1247 */ + { 0x0000, 0x0000, 0x0000 }, /* R1248 */ + { 0x0000, 0x0000, 0x0000 }, /* R1249 */ + { 0x0000, 0x0000, 0x0000 }, /* R1250 */ + { 0x0000, 0x0000, 0x0000 }, /* R1251 */ + { 0x0000, 0x0000, 0x0000 }, /* R1252 */ + { 0x0000, 0x0000, 0x0000 }, /* R1253 */ + { 0x0000, 0x0000, 0x0000 }, /* R1254 */ + { 0x0000, 0x0000, 0x0000 }, /* R1255 */ + { 0x0000, 0x0000, 0x0000 }, /* R1256 */ + { 0x0000, 0x0000, 0x0000 }, /* R1257 */ + { 0x0000, 0x0000, 0x0000 }, /* R1258 */ + { 0x0000, 0x0000, 0x0000 }, /* R1259 */ + { 0x0000, 0x0000, 0x0000 }, /* R1260 */ + { 0x0000, 0x0000, 0x0000 }, /* R1261 */ + { 0x0000, 0x0000, 0x0000 }, /* R1262 */ + { 0x0000, 0x0000, 0x0000 }, /* R1263 */ + { 0x0000, 0x0000, 0x0000 }, /* R1264 */ + { 0x0000, 0x0000, 0x0000 }, /* R1265 */ + { 0x0000, 0x0000, 0x0000 }, /* R1266 */ + { 0x0000, 0x0000, 0x0000 }, /* R1267 */ + { 0x0000, 0x0000, 0x0000 }, /* R1268 */ + { 0x0000, 0x0000, 0x0000 }, /* R1269 */ + { 0x0000, 0x0000, 0x0000 }, /* R1270 */ + { 0x0000, 0x0000, 0x0000 }, /* R1271 */ + { 0x0000, 0x0000, 0x0000 }, /* R1272 */ + { 0x0000, 0x0000, 0x0000 }, /* R1273 */ + { 0x0000, 0x0000, 0x0000 }, /* R1274 */ + { 0x0000, 0x0000, 0x0000 }, /* R1275 */ + { 0x0000, 0x0000, 0x0000 }, /* R1276 */ + { 0x0000, 0x0000, 0x0000 }, /* R1277 */ + { 0x0000, 0x0000, 0x0000 }, /* R1278 */ + { 0x0000, 0x0000, 0x0000 }, /* R1279 */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1280 - AIF2 ADC Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1281 - AIF2 ADC Right Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1282 - AIF2 DAC Left Volume */ + { 0x00FF, 0x01FF, 0x0000 }, /* R1283 - AIF2 DAC Right Volume */ + { 0x0000, 0x0000, 0x0000 }, /* R1284 */ + { 0x0000, 0x0000, 0x0000 }, /* R1285 */ + { 0x0000, 0x0000, 0x0000 }, /* R1286 */ + { 0x0000, 0x0000, 0x0000 }, /* R1287 */ + { 0x0000, 0x0000, 0x0000 }, /* R1288 */ + { 0x0000, 0x0000, 0x0000 }, /* R1289 */ + { 0x0000, 0x0000, 0x0000 }, /* R1290 */ + { 0x0000, 0x0000, 0x0000 }, /* R1291 */ + { 0x0000, 0x0000, 0x0000 }, /* R1292 */ + { 0x0000, 0x0000, 0x0000 }, /* R1293 */ + { 0x0000, 0x0000, 0x0000 }, /* R1294 */ + { 0x0000, 0x0000, 0x0000 }, /* R1295 */ + { 0xF800, 0xF800, 0x0000 }, /* R1296 - AIF2 ADC Filters */ + { 0x0000, 0x0000, 0x0000 }, /* R1297 */ + { 0x0000, 0x0000, 0x0000 }, /* R1298 */ + { 0x0000, 0x0000, 0x0000 }, /* R1299 */ + { 0x0000, 0x0000, 0x0000 }, /* R1300 */ + { 0x0000, 0x0000, 0x0000 }, /* R1301 */ + { 0x0000, 0x0000, 0x0000 }, /* R1302 */ + { 0x0000, 0x0000, 0x0000 }, /* R1303 */ + { 0x0000, 0x0000, 0x0000 }, /* R1304 */ + { 0x0000, 0x0000, 0x0000 }, /* R1305 */ + { 0x0000, 0x0000, 0x0000 }, /* R1306 */ + { 0x0000, 0x0000, 0x0000 }, /* R1307 */ + { 0x0000, 0x0000, 0x0000 }, /* R1308 */ + { 0x0000, 0x0000, 0x0000 }, /* R1309 */ + { 0x0000, 0x0000, 0x0000 }, /* R1310 */ + { 0x0000, 0x0000, 0x0000 }, /* R1311 */ + { 0x02B6, 0x02B6, 0x0000 }, /* R1312 - AIF2 DAC Filters (1) */ + { 0x3F00, 0x3F00, 0x0000 }, /* R1313 - AIF2 DAC Filters (2) */ + { 0x0000, 0x0000, 0x0000 }, /* R1314 */ + { 0x0000, 0x0000, 0x0000 }, /* R1315 */ + { 0x0000, 0x0000, 0x0000 }, /* R1316 */ + { 0x0000, 0x0000, 0x0000 }, /* R1317 */ + { 0x0000, 0x0000, 0x0000 }, /* R1318 */ + { 0x0000, 0x0000, 0x0000 }, /* R1319 */ + { 0x0000, 0x0000, 0x0000 }, /* R1320 */ + { 0x0000, 0x0000, 0x0000 }, /* R1321 */ + { 0x0000, 0x0000, 0x0000 }, /* R1322 */ + { 0x0000, 0x0000, 0x0000 }, /* R1323 */ + { 0x0000, 0x0000, 0x0000 }, /* R1324 */ + { 0x0000, 0x0000, 0x0000 }, /* R1325 */ + { 0x0000, 0x0000, 0x0000 }, /* R1326 */ + { 0x0000, 0x0000, 0x0000 }, /* R1327 */ + { 0x0000, 0x0000, 0x0000 }, /* R1328 */ + { 0x0000, 0x0000, 0x0000 }, /* R1329 */ + { 0x0000, 0x0000, 0x0000 }, /* R1330 */ + { 0x0000, 0x0000, 0x0000 }, /* R1331 */ + { 0x0000, 0x0000, 0x0000 }, /* R1332 */ + { 0x0000, 0x0000, 0x0000 }, /* R1333 */ + { 0x0000, 0x0000, 0x0000 }, /* R1334 */ + { 0x0000, 0x0000, 0x0000 }, /* R1335 */ + { 0x0000, 0x0000, 0x0000 }, /* R1336 */ + { 0x0000, 0x0000, 0x0000 }, /* R1337 */ + { 0x0000, 0x0000, 0x0000 }, /* R1338 */ + { 0x0000, 0x0000, 0x0000 }, /* R1339 */ + { 0x0000, 0x0000, 0x0000 }, /* R1340 */ + { 0x0000, 0x0000, 0x0000 }, /* R1341 */ + { 0x0000, 0x0000, 0x0000 }, /* R1342 */ + { 0x0000, 0x0000, 0x0000 }, /* R1343 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1344 - AIF2 DRC (1) */ + { 0x1FFF, 0x1FFF, 0x0000 }, /* R1345 - AIF2 DRC (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1346 - AIF2 DRC (3) */ + { 0x07FF, 0x07FF, 0x0000 }, /* R1347 - AIF2 DRC (4) */ + { 0x03FF, 0x03FF, 0x0000 }, /* R1348 - AIF2 DRC (5) */ + { 0x0000, 0x0000, 0x0000 }, /* R1349 */ + { 0x0000, 0x0000, 0x0000 }, /* R1350 */ + { 0x0000, 0x0000, 0x0000 }, /* R1351 */ + { 0x0000, 0x0000, 0x0000 }, /* R1352 */ + { 0x0000, 0x0000, 0x0000 }, /* R1353 */ + { 0x0000, 0x0000, 0x0000 }, /* R1354 */ + { 0x0000, 0x0000, 0x0000 }, /* R1355 */ + { 0x0000, 0x0000, 0x0000 }, /* R1356 */ + { 0x0000, 0x0000, 0x0000 }, /* R1357 */ + { 0x0000, 0x0000, 0x0000 }, /* R1358 */ + { 0x0000, 0x0000, 0x0000 }, /* R1359 */ + { 0x0000, 0x0000, 0x0000 }, /* R1360 */ + { 0x0000, 0x0000, 0x0000 }, /* R1361 */ + { 0x0000, 0x0000, 0x0000 }, /* R1362 */ + { 0x0000, 0x0000, 0x0000 }, /* R1363 */ + { 0x0000, 0x0000, 0x0000 }, /* R1364 */ + { 0x0000, 0x0000, 0x0000 }, /* R1365 */ + { 0x0000, 0x0000, 0x0000 }, /* R1366 */ + { 0x0000, 0x0000, 0x0000 }, /* R1367 */ + { 0x0000, 0x0000, 0x0000 }, /* R1368 */ + { 0x0000, 0x0000, 0x0000 }, /* R1369 */ + { 0x0000, 0x0000, 0x0000 }, /* R1370 */ + { 0x0000, 0x0000, 0x0000 }, /* R1371 */ + { 0x0000, 0x0000, 0x0000 }, /* R1372 */ + { 0x0000, 0x0000, 0x0000 }, /* R1373 */ + { 0x0000, 0x0000, 0x0000 }, /* R1374 */ + { 0x0000, 0x0000, 0x0000 }, /* R1375 */ + { 0x0000, 0x0000, 0x0000 }, /* R1376 */ + { 0x0000, 0x0000, 0x0000 }, /* R1377 */ + { 0x0000, 0x0000, 0x0000 }, /* R1378 */ + { 0x0000, 0x0000, 0x0000 }, /* R1379 */ + { 0x0000, 0x0000, 0x0000 }, /* R1380 */ + { 0x0000, 0x0000, 0x0000 }, /* R1381 */ + { 0x0000, 0x0000, 0x0000 }, /* R1382 */ + { 0x0000, 0x0000, 0x0000 }, /* R1383 */ + { 0x0000, 0x0000, 0x0000 }, /* R1384 */ + { 0x0000, 0x0000, 0x0000 }, /* R1385 */ + { 0x0000, 0x0000, 0x0000 }, /* R1386 */ + { 0x0000, 0x0000, 0x0000 }, /* R1387 */ + { 0x0000, 0x0000, 0x0000 }, /* R1388 */ + { 0x0000, 0x0000, 0x0000 }, /* R1389 */ + { 0x0000, 0x0000, 0x0000 }, /* R1390 */ + { 0x0000, 0x0000, 0x0000 }, /* R1391 */ + { 0x0000, 0x0000, 0x0000 }, /* R1392 */ + { 0x0000, 0x0000, 0x0000 }, /* R1393 */ + { 0x0000, 0x0000, 0x0000 }, /* R1394 */ + { 0x0000, 0x0000, 0x0000 }, /* R1395 */ + { 0x0000, 0x0000, 0x0000 }, /* R1396 */ + { 0x0000, 0x0000, 0x0000 }, /* R1397 */ + { 0x0000, 0x0000, 0x0000 }, /* R1398 */ + { 0x0000, 0x0000, 0x0000 }, /* R1399 */ + { 0x0000, 0x0000, 0x0000 }, /* R1400 */ + { 0x0000, 0x0000, 0x0000 }, /* R1401 */ + { 0x0000, 0x0000, 0x0000 }, /* R1402 */ + { 0x0000, 0x0000, 0x0000 }, /* R1403 */ + { 0x0000, 0x0000, 0x0000 }, /* R1404 */ + { 0x0000, 0x0000, 0x0000 }, /* R1405 */ + { 0x0000, 0x0000, 0x0000 }, /* R1406 */ + { 0x0000, 0x0000, 0x0000 }, /* R1407 */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1408 - AIF2 EQ Gains (1) */ + { 0xFFC0, 0xFFC0, 0x0000 }, /* R1409 - AIF2 EQ Gains (2) */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1410 - AIF2 EQ Band 1 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1411 - AIF2 EQ Band 1 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1412 - AIF2 EQ Band 1 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1413 - AIF2 EQ Band 2 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1414 - AIF2 EQ Band 2 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1415 - AIF2 EQ Band 2 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1416 - AIF2 EQ Band 2 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1417 - AIF2 EQ Band 3 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1418 - AIF2 EQ Band 3 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1419 - AIF2 EQ Band 3 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1420 - AIF2 EQ Band 3 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1421 - AIF2 EQ Band 4 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1422 - AIF2 EQ Band 4 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1423 - AIF2 EQ Band 4 C */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1424 - AIF2 EQ Band 4 PG */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1425 - AIF2 EQ Band 5 A */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1426 - AIF2 EQ Band 5 B */ + { 0xFFFF, 0xFFFF, 0x0000 }, /* R1427 - AIF2 EQ Band 5 PG */ + { 0x0000, 0x0000, 0x0000 }, /* R1428 */ + { 0x0000, 0x0000, 0x0000 }, /* R1429 */ + { 0x0000, 0x0000, 0x0000 }, /* R1430 */ + { 0x0000, 0x0000, 0x0000 }, /* R1431 */ + { 0x0000, 0x0000, 0x0000 }, /* R1432 */ + { 0x0000, 0x0000, 0x0000 }, /* R1433 */ + { 0x0000, 0x0000, 0x0000 }, /* R1434 */ + { 0x0000, 0x0000, 0x0000 }, /* R1435 */ + { 0x0000, 0x0000, 0x0000 }, /* R1436 */ + { 0x0000, 0x0000, 0x0000 }, /* R1437 */ + { 0x0000, 0x0000, 0x0000 }, /* R1438 */ + { 0x0000, 0x0000, 0x0000 }, /* R1439 */ + { 0x0000, 0x0000, 0x0000 }, /* R1440 */ + { 0x0000, 0x0000, 0x0000 }, /* R1441 */ + { 0x0000, 0x0000, 0x0000 }, /* R1442 */ + { 0x0000, 0x0000, 0x0000 }, /* R1443 */ + { 0x0000, 0x0000, 0x0000 }, /* R1444 */ + { 0x0000, 0x0000, 0x0000 }, /* R1445 */ + { 0x0000, 0x0000, 0x0000 }, /* R1446 */ + { 0x0000, 0x0000, 0x0000 }, /* R1447 */ + { 0x0000, 0x0000, 0x0000 }, /* R1448 */ + { 0x0000, 0x0000, 0x0000 }, /* R1449 */ + { 0x0000, 0x0000, 0x0000 }, /* R1450 */ + { 0x0000, 0x0000, 0x0000 }, /* R1451 */ + { 0x0000, 0x0000, 0x0000 }, /* R1452 */ + { 0x0000, 0x0000, 0x0000 }, /* R1453 */ + { 0x0000, 0x0000, 0x0000 }, /* R1454 */ + { 0x0000, 0x0000, 0x0000 }, /* R1455 */ + { 0x0000, 0x0000, 0x0000 }, /* R1456 */ + { 0x0000, 0x0000, 0x0000 }, /* R1457 */ + { 0x0000, 0x0000, 0x0000 }, /* R1458 */ + { 0x0000, 0x0000, 0x0000 }, /* R1459 */ + { 0x0000, 0x0000, 0x0000 }, /* R1460 */ + { 0x0000, 0x0000, 0x0000 }, /* R1461 */ + { 0x0000, 0x0000, 0x0000 }, /* R1462 */ + { 0x0000, 0x0000, 0x0000 }, /* R1463 */ + { 0x0000, 0x0000, 0x0000 }, /* R1464 */ + { 0x0000, 0x0000, 0x0000 }, /* R1465 */ + { 0x0000, 0x0000, 0x0000 }, /* R1466 */ + { 0x0000, 0x0000, 0x0000 }, /* R1467 */ + { 0x0000, 0x0000, 0x0000 }, /* R1468 */ + { 0x0000, 0x0000, 0x0000 }, /* R1469 */ + { 0x0000, 0x0000, 0x0000 }, /* R1470 */ + { 0x0000, 0x0000, 0x0000 }, /* R1471 */ + { 0x0000, 0x0000, 0x0000 }, /* R1472 */ + { 0x0000, 0x0000, 0x0000 }, /* R1473 */ + { 0x0000, 0x0000, 0x0000 }, /* R1474 */ + { 0x0000, 0x0000, 0x0000 }, /* R1475 */ + { 0x0000, 0x0000, 0x0000 }, /* R1476 */ + { 0x0000, 0x0000, 0x0000 }, /* R1477 */ + { 0x0000, 0x0000, 0x0000 }, /* R1478 */ + { 0x0000, 0x0000, 0x0000 }, /* R1479 */ + { 0x0000, 0x0000, 0x0000 }, /* R1480 */ + { 0x0000, 0x0000, 0x0000 }, /* R1481 */ + { 0x0000, 0x0000, 0x0000 }, /* R1482 */ + { 0x0000, 0x0000, 0x0000 }, /* R1483 */ + { 0x0000, 0x0000, 0x0000 }, /* R1484 */ + { 0x0000, 0x0000, 0x0000 }, /* R1485 */ + { 0x0000, 0x0000, 0x0000 }, /* R1486 */ + { 0x0000, 0x0000, 0x0000 }, /* R1487 */ + { 0x0000, 0x0000, 0x0000 }, /* R1488 */ + { 0x0000, 0x0000, 0x0000 }, /* R1489 */ + { 0x0000, 0x0000, 0x0000 }, /* R1490 */ + { 0x0000, 0x0000, 0x0000 }, /* R1491 */ + { 0x0000, 0x0000, 0x0000 }, /* R1492 */ + { 0x0000, 0x0000, 0x0000 }, /* R1493 */ + { 0x0000, 0x0000, 0x0000 }, /* R1494 */ + { 0x0000, 0x0000, 0x0000 }, /* R1495 */ + { 0x0000, 0x0000, 0x0000 }, /* R1496 */ + { 0x0000, 0x0000, 0x0000 }, /* R1497 */ + { 0x0000, 0x0000, 0x0000 }, /* R1498 */ + { 0x0000, 0x0000, 0x0000 }, /* R1499 */ + { 0x0000, 0x0000, 0x0000 }, /* R1500 */ + { 0x0000, 0x0000, 0x0000 }, /* R1501 */ + { 0x0000, 0x0000, 0x0000 }, /* R1502 */ + { 0x0000, 0x0000, 0x0000 }, /* R1503 */ + { 0x0000, 0x0000, 0x0000 }, /* R1504 */ + { 0x0000, 0x0000, 0x0000 }, /* R1505 */ + { 0x0000, 0x0000, 0x0000 }, /* R1506 */ + { 0x0000, 0x0000, 0x0000 }, /* R1507 */ + { 0x0000, 0x0000, 0x0000 }, /* R1508 */ + { 0x0000, 0x0000, 0x0000 }, /* R1509 */ + { 0x0000, 0x0000, 0x0000 }, /* R1510 */ + { 0x0000, 0x0000, 0x0000 }, /* R1511 */ + { 0x0000, 0x0000, 0x0000 }, /* R1512 */ + { 0x0000, 0x0000, 0x0000 }, /* R1513 */ + { 0x0000, 0x0000, 0x0000 }, /* R1514 */ + { 0x0000, 0x0000, 0x0000 }, /* R1515 */ + { 0x0000, 0x0000, 0x0000 }, /* R1516 */ + { 0x0000, 0x0000, 0x0000 }, /* R1517 */ + { 0x0000, 0x0000, 0x0000 }, /* R1518 */ + { 0x0000, 0x0000, 0x0000 }, /* R1519 */ + { 0x0000, 0x0000, 0x0000 }, /* R1520 */ + { 0x0000, 0x0000, 0x0000 }, /* R1521 */ + { 0x0000, 0x0000, 0x0000 }, /* R1522 */ + { 0x0000, 0x0000, 0x0000 }, /* R1523 */ + { 0x0000, 0x0000, 0x0000 }, /* R1524 */ + { 0x0000, 0x0000, 0x0000 }, /* R1525 */ + { 0x0000, 0x0000, 0x0000 }, /* R1526 */ + { 0x0000, 0x0000, 0x0000 }, /* R1527 */ + { 0x0000, 0x0000, 0x0000 }, /* R1528 */ + { 0x0000, 0x0000, 0x0000 }, /* R1529 */ + { 0x0000, 0x0000, 0x0000 }, /* R1530 */ + { 0x0000, 0x0000, 0x0000 }, /* R1531 */ + { 0x0000, 0x0000, 0x0000 }, /* R1532 */ + { 0x0000, 0x0000, 0x0000 }, /* R1533 */ + { 0x0000, 0x0000, 0x0000 }, /* R1534 */ + { 0x0000, 0x0000, 0x0000 }, /* R1535 */ + { 0x01EF, 0x01EF, 0x0000 }, /* R1536 - DAC1 Mixer Volumes */ + { 0x0037, 0x0037, 0x0000 }, /* R1537 - DAC1 Left Mixer Routing */ + { 0x0037, 0x0037, 0x0000 }, /* R1538 - DAC1 Right Mixer Routing */ + { 0x01EF, 0x01EF, 0x0000 }, /* R1539 - DAC2 Mixer Volumes */ + { 0x0037, 0x0037, 0x0000 }, /* R1540 - DAC2 Left Mixer Routing */ + { 0x0037, 0x0037, 0x0000 }, /* R1541 - DAC2 Right Mixer Routing */ + { 0x0003, 0x0003, 0x0000 }, /* R1542 - AIF1 ADC1 Left Mixer Routing */ + { 0x0003, 0x0003, 0x0000 }, /* R1543 - AIF1 ADC1 Right Mixer Routing */ + { 0x0003, 0x0003, 0x0000 }, /* R1544 - AIF1 ADC2 Left Mixer Routing */ + { 0x0003, 0x0003, 0x0000 }, /* R1545 - AIF1 ADC2 Right mixer Routing */ + { 0x0000, 0x0000, 0x0000 }, /* R1546 */ + { 0x0000, 0x0000, 0x0000 }, /* R1547 */ + { 0x0000, 0x0000, 0x0000 }, /* R1548 */ + { 0x0000, 0x0000, 0x0000 }, /* R1549 */ + { 0x0000, 0x0000, 0x0000 }, /* R1550 */ + { 0x0000, 0x0000, 0x0000 }, /* R1551 */ + { 0x02FF, 0x03FF, 0x0000 }, /* R1552 - DAC1 Left Volume */ + { 0x02FF, 0x03FF, 0x0000 }, /* R1553 - DAC1 Right Volume */ + { 0x02FF, 0x03FF, 0x0000 }, /* R1554 - DAC2 Left Volume */ + { 0x02FF, 0x03FF, 0x0000 }, /* R1555 - DAC2 Right Volume */ + { 0x0003, 0x0003, 0x0000 }, /* R1556 - DAC Softmute */ + { 0x0000, 0x0000, 0x0000 }, /* R1557 */ + { 0x0000, 0x0000, 0x0000 }, /* R1558 */ + { 0x0000, 0x0000, 0x0000 }, /* R1559 */ + { 0x0000, 0x0000, 0x0000 }, /* R1560 */ + { 0x0000, 0x0000, 0x0000 }, /* R1561 */ + { 0x0000, 0x0000, 0x0000 }, /* R1562 */ + { 0x0000, 0x0000, 0x0000 }, /* R1563 */ + { 0x0000, 0x0000, 0x0000 }, /* R1564 */ + { 0x0000, 0x0000, 0x0000 }, /* R1565 */ + { 0x0000, 0x0000, 0x0000 }, /* R1566 */ + { 0x0000, 0x0000, 0x0000 }, /* R1567 */ + { 0x0003, 0x0003, 0x0000 }, /* R1568 - Oversampling */ + { 0x03C3, 0x03C3, 0x0000 }, /* R1569 - Sidetone */ +}; + +static int wm8994_readable(unsigned int reg) +{ + if (reg >= ARRAY_SIZE(access_masks)) + return 0; + return access_masks[reg].readable != 0; +} + +static int wm8994_volatile(unsigned int reg) +{ + if (reg >= WM8994_REG_CACHE_SIZE) + return 1; + + switch (reg) { + case WM8994_SOFTWARE_RESET: + case WM8994_CHIP_REVISION: + case WM8994_DC_SERVO_1: + case WM8994_DC_SERVO_READBACK: + case WM8994_RATE_STATUS: + case WM8994_LDO_1: + case WM8994_LDO_2: + return 1; + default: + return 0; + } +} + +static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct wm8994_priv *wm8994 = codec->private_data; + + BUG_ON(reg > WM8994_MAX_REGISTER); + + if (!wm8994_volatile(reg)) + wm8994->reg_cache[reg] = value; + + return wm8994_reg_write(codec->control_data, reg, value); +} + +static unsigned int wm8994_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *reg_cache = codec->reg_cache; + + BUG_ON(reg > WM8994_MAX_REGISTER); + + if (wm8994_volatile(reg)) + return wm8994_reg_read(codec->control_data, reg); + else + return reg_cache[reg]; +} + +static int configure_aif_clock(struct snd_soc_codec *codec, int aif) +{ + struct wm8994_priv *wm8994 = codec->private_data; + int rate; + int reg1 = 0; + int offset; + + if (aif) + offset = 4; + else + offset = 0; + + switch (wm8994->sysclk[aif]) { + case WM8994_SYSCLK_MCLK1: + rate = wm8994->mclk[0]; + break; + + case WM8994_SYSCLK_MCLK2: + reg1 |= 0x8; + rate = wm8994->mclk[1]; + break; + + case WM8994_SYSCLK_FLL1: + reg1 |= 0x10; + rate = wm8994->fll[0].out; + break; + + case WM8994_SYSCLK_FLL2: + reg1 |= 0x18; + rate = wm8994->fll[1].out; + break; + + default: + return -EINVAL; + } + + if (rate >= 13500000) { + rate /= 2; + reg1 |= WM8994_AIF1CLK_DIV; + + dev_dbg(codec->dev, "Dividing AIF%d clock to %dHz\n", + aif + 1, rate); + } + wm8994->aifclk[aif] = rate; + + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset, + WM8994_AIF1CLK_SRC_MASK | WM8994_AIF1CLK_DIV, + reg1); + + return 0; +} + +static int configure_clock(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = codec->private_data; + int old, new; + + /* Bring up the AIF clocks first */ + configure_aif_clock(codec, 0); + configure_aif_clock(codec, 1); + + /* Then switch CLK_SYS over to the higher of them; a change + * can only happen as a result of a clocking change which can + * only be made outside of DAPM so we can safely redo the + * clocking. + */ + + /* If they're equal it doesn't matter which is used */ + if (wm8994->aifclk[0] == wm8994->aifclk[1]) + return 0; + + if (wm8994->aifclk[0] < wm8994->aifclk[1]) + new = WM8994_SYSCLK_SRC; + else + new = 0; + + old = snd_soc_read(codec, WM8994_CLOCKING_1) & WM8994_SYSCLK_SRC; + + /* If there's no change then we're done. */ + if (old == new) + return 0; + + snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static int check_clk_sys(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + int reg = snd_soc_read(source->codec, WM8994_CLOCKING_1); + const char *clk; + + /* Check what we're currently using for CLK_SYS */ + if (reg & WM8994_SYSCLK_SRC) + clk = "AIF2CLK"; + else + clk = "AIF1CLK"; + + return strcmp(source->name, clk) == 0; +} + +static const char *sidetone_hpf_text[] = { + "2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz" +}; + +static const struct soc_enum sidetone_hpf = + SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text); + +static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); +static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); + +#define WM8994_DRC_SWITCH(xname, reg, shift) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ + .put = wm8994_put_drc_sw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, 1, 0) } + +static int wm8994_put_drc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int mask, ret; + + /* Can't enable both ADC and DAC paths simultaneously */ + if (mc->shift == WM8994_AIF1DAC1_DRC_ENA_SHIFT) + mask = WM8994_AIF1ADC1L_DRC_ENA_MASK | + WM8994_AIF1ADC1R_DRC_ENA_MASK; + else + mask = WM8994_AIF1DAC1_DRC_ENA_MASK; + + ret = snd_soc_read(codec, mc->reg); + if (ret < 0) + return ret; + if (ret & mask) + return -EINVAL; + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + + + +static void wm8994_set_drc(struct snd_soc_codec *codec, int drc) +{ + struct wm8994_priv *wm8994 = codec->private_data; + struct wm8994_pdata *pdata = wm8994->pdata; + int base = wm8994_drc_base[drc]; + int cfg = wm8994->drc_cfg[drc]; + int save, i; + + /* Save any enables; the configuration should clear them. */ + save = snd_soc_read(codec, base); + save &= WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA | + WM8994_AIF1ADC1R_DRC_ENA; + + for (i = 0; i < WM8994_DRC_REGS; i++) + snd_soc_update_bits(codec, base + i, 0xffff, + pdata->drc_cfgs[cfg].regs[i]); + + snd_soc_update_bits(codec, base, WM8994_AIF1DAC1_DRC_ENA | + WM8994_AIF1ADC1L_DRC_ENA | + WM8994_AIF1ADC1R_DRC_ENA, save); +} + +/* Icky as hell but saves code duplication */ +static int wm8994_get_drc(const char *name) +{ + if (strcmp(name, "AIF1DRC1 Mode") == 0) + return 0; + if (strcmp(name, "AIF1DRC2 Mode") == 0) + return 1; + if (strcmp(name, "AIF2DRC Mode") == 0) + return 2; + return -EINVAL; +} + +static int wm8994_put_drc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = codec->private_data; + struct wm8994_pdata *pdata = wm8994->pdata; + int drc = wm8994_get_drc(kcontrol->id.name); + int value = ucontrol->value.integer.value[0]; + + if (drc < 0) + return drc; + + if (value >= pdata->num_drc_cfgs) + return -EINVAL; + + wm8994->drc_cfg[drc] = value; + + wm8994_set_drc(codec, drc); + + return 0; +} + +static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = codec->private_data; + int drc = wm8994_get_drc(kcontrol->id.name); + + ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc]; + + return 0; +} + +static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block) +{ + struct wm8994_priv *wm8994 = codec->private_data; + struct wm8994_pdata *pdata = wm8994->pdata; + int base = wm8994_retune_mobile_base[block]; + int iface, best, best_val, save, i, cfg; + + if (!pdata || !wm8994->num_retune_mobile_texts) + return; + + switch (block) { + case 0: + case 1: + iface = 0; + break; + case 2: + iface = 1; + break; + default: + return; + } + + /* Find the version of the currently selected configuration + * with the nearest sample rate. */ + cfg = wm8994->retune_mobile_cfg[block]; + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8994->retune_mobile_texts[cfg]) == 0 && + abs(pdata->retune_mobile_cfgs[i].rate + - wm8994->dac_rates[iface]) < best_val) { + best = i; + best_val = abs(pdata->retune_mobile_cfgs[i].rate + - wm8994->dac_rates[iface]); + } + } + + dev_dbg(codec->dev, "ReTune Mobile %d %s/%dHz for %dHz sample rate\n", + block, + pdata->retune_mobile_cfgs[best].name, + pdata->retune_mobile_cfgs[best].rate, + wm8994->dac_rates[iface]); + + /* The EQ will be disabled while reconfiguring it, remember the + * current configuration. + */ + save = snd_soc_read(codec, base); + save &= WM8994_AIF1DAC1_EQ_ENA; + + for (i = 0; i < WM8994_EQ_REGS; i++) + snd_soc_update_bits(codec, base + i, 0xffff, + pdata->retune_mobile_cfgs[best].regs[i]); + + snd_soc_update_bits(codec, base, WM8994_AIF1DAC1_EQ_ENA, save); +} + +/* Icky as hell but saves code duplication */ +static int wm8994_get_retune_mobile_block(const char *name) +{ + if (strcmp(name, "AIF1.1 EQ Mode") == 0) + return 0; + if (strcmp(name, "AIF1.2 EQ Mode") == 0) + return 1; + if (strcmp(name, "AIF2 EQ Mode") == 0) + return 2; + return -EINVAL; +} + +static int wm8994_put_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = codec->private_data; + struct wm8994_pdata *pdata = wm8994->pdata; + int block = wm8994_get_retune_mobile_block(kcontrol->id.name); + int value = ucontrol->value.integer.value[0]; + + if (block < 0) + return block; + + if (value >= pdata->num_retune_mobile_cfgs) + return -EINVAL; + + wm8994->retune_mobile_cfg[block] = value; + + wm8994_set_retune_mobile(codec, block); + + return 0; +} + +static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = codec->private_data; + int block = wm8994_get_retune_mobile_block(kcontrol->id.name); + + ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block]; + + return 0; +} + +static const struct snd_kcontrol_new wm8994_snd_controls[] = { +SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, + WM8994_AIF1_ADC1_RIGHT_VOLUME, + 1, 119, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME, + WM8994_AIF1_ADC2_RIGHT_VOLUME, + 1, 119, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME, + WM8994_AIF2_ADC_RIGHT_VOLUME, + 1, 119, 0, digital_tlv), + +SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME, + WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME, + WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF2DAC Volume", WM8994_AIF2_DAC_LEFT_VOLUME, + WM8994_AIF2_DAC_RIGHT_VOLUME, 1, 96, 0, digital_tlv), + +SOC_SINGLE_TLV("AIF1 Boost Volume", WM8994_AIF1_CONTROL_2, 10, 3, 0, aif_tlv), +SOC_SINGLE_TLV("AIF2 Boost Volume", WM8994_AIF2_CONTROL_2, 10, 3, 0, aif_tlv), + +SOC_SINGLE("AIF1DAC1 EQ Switch", WM8994_AIF1_DAC1_EQ_GAINS_1, 0, 1, 0), +SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0), +SOC_SINGLE("AIF2 EQ Switch", WM8994_AIF2_EQ_GAINS_1, 0, 1, 0), + +WM8994_DRC_SWITCH("AIF1DAC1 DRC Switch", WM8994_AIF1_DRC1_1, 2), +WM8994_DRC_SWITCH("AIF1ADC1L DRC Switch", WM8994_AIF1_DRC1_1, 1), +WM8994_DRC_SWITCH("AIF1ADC1R DRC Switch", WM8994_AIF1_DRC1_1, 0), + +WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2), +WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1), +WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0), + +WM8994_DRC_SWITCH("AIF2DAC DRC Switch", WM8994_AIF2_DRC_1, 2), +WM8994_DRC_SWITCH("AIF2ADCL DRC Switch", WM8994_AIF2_DRC_1, 1), +WM8994_DRC_SWITCH("AIF2ADCR DRC Switch", WM8994_AIF2_DRC_1, 0), + +SOC_SINGLE_TLV("DAC1 Right Sidetone Volume", WM8994_DAC1_MIXER_VOLUMES, + 5, 12, 0, st_tlv), +SOC_SINGLE_TLV("DAC1 Left Sidetone Volume", WM8994_DAC1_MIXER_VOLUMES, + 0, 12, 0, st_tlv), +SOC_SINGLE_TLV("DAC2 Right Sidetone Volume", WM8994_DAC2_MIXER_VOLUMES, + 5, 12, 0, st_tlv), +SOC_SINGLE_TLV("DAC2 Left Sidetone Volume", WM8994_DAC2_MIXER_VOLUMES, + 0, 12, 0, st_tlv), +SOC_ENUM("Sidetone HPF Mux", sidetone_hpf), +SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0), + +SOC_DOUBLE_R_TLV("DAC1 Volume", WM8994_DAC1_LEFT_VOLUME, + WM8994_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R("DAC1 Switch", WM8994_DAC1_LEFT_VOLUME, + WM8994_DAC1_RIGHT_VOLUME, 9, 1, 1), + +SOC_DOUBLE_R_TLV("DAC2 Volume", WM8994_DAC2_LEFT_VOLUME, + WM8994_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R("DAC2 Switch", WM8994_DAC2_LEFT_VOLUME, + WM8994_DAC2_RIGHT_VOLUME, 9, 1, 1), + +SOC_SINGLE_TLV("SPKL DAC2 Volume", WM8994_SPKMIXL_ATTENUATION, + 6, 1, 1, wm_hubs_spkmix_tlv), +SOC_SINGLE_TLV("SPKL DAC1 Volume", WM8994_SPKMIXL_ATTENUATION, + 2, 1, 1, wm_hubs_spkmix_tlv), + +SOC_SINGLE_TLV("SPKR DAC2 Volume", WM8994_SPKMIXR_ATTENUATION, + 6, 1, 1, wm_hubs_spkmix_tlv), +SOC_SINGLE_TLV("SPKR DAC1 Volume", WM8994_SPKMIXR_ATTENUATION, + 2, 1, 1, wm_hubs_spkmix_tlv), + +SOC_SINGLE_TLV("AIF1DAC1 3D Stereo Volume", WM8994_AIF1_DAC1_FILTERS_2, + 10, 15, 0, wm8994_3d_tlv), +SOC_SINGLE("AIF1DAC1 3D Stereo Switch", WM8994_AIF1_DAC2_FILTERS_2, + 8, 1, 0), +SOC_SINGLE_TLV("AIF1DAC2 3D Stereo Volume", WM8994_AIF1_DAC2_FILTERS_2, + 10, 15, 0, wm8994_3d_tlv), +SOC_SINGLE("AIF1DAC2 3D Stereo Switch", WM8994_AIF1_DAC2_FILTERS_2, + 8, 1, 0), +SOC_SINGLE_TLV("AIF2DAC 3D Stereo Volume", WM8994_AIF1_DAC1_FILTERS_2, + 10, 15, 0, wm8994_3d_tlv), +SOC_SINGLE("AIF2DAC 3D Stereo Switch", WM8994_AIF1_DAC2_FILTERS_2, + 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8994_eq_controls[] = { +SOC_SINGLE_TLV("AIF1DAC1 EQ1 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC1 EQ2 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 6, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC1 EQ3 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 1, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC1 EQ4 Volume", WM8994_AIF1_DAC1_EQ_GAINS_2, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC1 EQ5 Volume", WM8994_AIF1_DAC1_EQ_GAINS_2, 6, 31, 0, + eq_tlv), + +SOC_SINGLE_TLV("AIF1DAC2 EQ1 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC2 EQ2 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 6, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC2 EQ3 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 1, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC2 EQ4 Volume", WM8994_AIF1_DAC2_EQ_GAINS_2, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF1DAC2 EQ5 Volume", WM8994_AIF1_DAC2_EQ_GAINS_2, 6, 31, 0, + eq_tlv), + +SOC_SINGLE_TLV("AIF2 EQ1 Volume", WM8994_AIF2_EQ_GAINS_1, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF2 EQ2 Volume", WM8994_AIF2_EQ_GAINS_1, 6, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF2 EQ3 Volume", WM8994_AIF2_EQ_GAINS_1, 1, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF2 EQ4 Volume", WM8994_AIF2_EQ_GAINS_2, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, + eq_tlv), +}; + +static int clk_sys_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return configure_clock(codec); + + case SND_SOC_DAPM_POST_PMD: + configure_clock(codec); + break; + } + + return 0; +} + +static void wm8994_update_class_w(struct snd_soc_codec *codec) +{ + int enable = 1; + int source = 0; /* GCC flow analysis can't track enable */ + int reg, reg_r; + + /* Only support direct DAC->headphone paths */ + reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_1); + if (!(reg & WM8994_DAC1L_TO_HPOUT1L)) { + dev_dbg(codec->dev, "HPL connected to output mixer\n"); + enable = 0; + } + + reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_2); + if (!(reg & WM8994_DAC1R_TO_HPOUT1R)) { + dev_dbg(codec->dev, "HPR connected to output mixer\n"); + enable = 0; + } + + /* We also need the same setting for L/R and only one path */ + reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING); + switch (reg) { + case WM8994_AIF2DACL_TO_DAC1L: + dev_dbg(codec->dev, "Class W source AIF2DAC\n"); + source = 2 << WM8994_CP_DYN_SRC_SEL_SHIFT; + break; + case WM8994_AIF1DAC2L_TO_DAC1L: + dev_dbg(codec->dev, "Class W source AIF1DAC2\n"); + source = 1 << WM8994_CP_DYN_SRC_SEL_SHIFT; + break; + case WM8994_AIF1DAC1L_TO_DAC1L: + dev_dbg(codec->dev, "Class W source AIF1DAC1\n"); + source = 0 << WM8994_CP_DYN_SRC_SEL_SHIFT; + break; + default: + dev_dbg(codec->dev, "DAC mixer setting: %x\n", reg); + enable = 0; + break; + } + + reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING); + if (reg_r != reg) { + dev_dbg(codec->dev, "Left and right DAC mixers different\n"); + enable = 0; + } + + if (enable) { + dev_dbg(codec->dev, "Class W enabled\n"); + snd_soc_update_bits(codec, WM8994_CLASS_W_1, + WM8994_CP_DYN_PWR | + WM8994_CP_DYN_SRC_SEL_MASK, + source | WM8994_CP_DYN_PWR); + + } else { + dev_dbg(codec->dev, "Class W disabled\n"); + snd_soc_update_bits(codec, WM8994_CLASS_W_1, + WM8994_CP_DYN_PWR, 0); + } +} + +static const char *hp_mux_text[] = { + "Mixer", + "DAC", +}; + +#define WM8994_HP_ENUM(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_double, \ + .put = wm8994_put_hp_enum, \ + .private_value = (unsigned long)&xenum } + +static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = w->codec; + int ret; + + ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + wm8994_update_class_w(codec); + + return ret; +} + +static const struct soc_enum hpl_enum = + SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_1, 8, 2, hp_mux_text); + +static const struct snd_kcontrol_new hpl_mux = + WM8994_HP_ENUM("Left Headphone Mux", hpl_enum); + +static const struct soc_enum hpr_enum = + SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_2, 8, 2, hp_mux_text); + +static const struct snd_kcontrol_new hpr_mux = + WM8994_HP_ENUM("Right Headphone Mux", hpr_enum); + +static const char *adc_mux_text[] = { + "ADC", + "DMIC", +}; + +static const struct soc_enum adc_enum = + SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text); + +static const struct snd_kcontrol_new adcl_mux = + SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); + +static const struct snd_kcontrol_new adcr_mux = + SOC_DAPM_ENUM_VIRT("ADCR Mux", adc_enum); + +static const struct snd_kcontrol_new left_speaker_mixer[] = { +SOC_DAPM_SINGLE("DAC2 Switch", WM8994_SPEAKER_MIXER, 9, 1, 0), +SOC_DAPM_SINGLE("Input Switch", WM8994_SPEAKER_MIXER, 7, 1, 0), +SOC_DAPM_SINGLE("IN1LP Switch", WM8994_SPEAKER_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Output Switch", WM8994_SPEAKER_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("DAC1 Switch", WM8994_SPEAKER_MIXER, 1, 1, 0), +}; + +static const struct snd_kcontrol_new right_speaker_mixer[] = { +SOC_DAPM_SINGLE("DAC2 Switch", WM8994_SPEAKER_MIXER, 8, 1, 0), +SOC_DAPM_SINGLE("Input Switch", WM8994_SPEAKER_MIXER, 6, 1, 0), +SOC_DAPM_SINGLE("IN1RP Switch", WM8994_SPEAKER_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Output Switch", WM8994_SPEAKER_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("DAC1 Switch", WM8994_SPEAKER_MIXER, 0, 1, 0), +}; + +/* Debugging; dump chip status after DAPM transitions */ +static int post_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + dev_dbg(codec->dev, "SRC status: %x\n", + snd_soc_read(codec, + WM8994_RATE_STATUS)); + return 0; +} + +static const struct snd_kcontrol_new aif1adc1l_mix[] = { +SOC_DAPM_SINGLE("ADC/DMIC Switch", WM8994_AIF1_ADC1_LEFT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new aif1adc1r_mix[] = { +SOC_DAPM_SINGLE("ADC/DMIC Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new aif2dac2l_mix[] = { +SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 5, 1, 0), +SOC_DAPM_SINGLE("Left Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 4, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 2, 1, 0), +SOC_DAPM_SINGLE("AIF1.2 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new aif2dac2r_mix[] = { +SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 5, 1, 0), +SOC_DAPM_SINGLE("Left Sidetone Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 4, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 2, 1, 0), +SOC_DAPM_SINGLE("AIF1.2 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + +#define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = wm8994_put_class_w, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = w->codec; + int ret; + + ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + wm8994_update_class_w(codec); + + return ret; +} + +static const struct snd_kcontrol_new dac1l_mix[] = { +WM8994_CLASS_W_SWITCH("Right Sidetone Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 5, 1, 0), +WM8994_CLASS_W_SWITCH("Left Sidetone Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 4, 1, 0), +WM8994_CLASS_W_SWITCH("AIF2 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 2, 1, 0), +WM8994_CLASS_W_SWITCH("AIF1.2 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 1, 1, 0), +WM8994_CLASS_W_SWITCH("AIF1.1 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new dac1r_mix[] = { +WM8994_CLASS_W_SWITCH("Right Sidetone Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 5, 1, 0), +WM8994_CLASS_W_SWITCH("Left Sidetone Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 4, 1, 0), +WM8994_CLASS_W_SWITCH("AIF2 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 2, 1, 0), +WM8994_CLASS_W_SWITCH("AIF1.2 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 1, 1, 0), +WM8994_CLASS_W_SWITCH("AIF1.1 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const char *sidetone_text[] = { + "ADC/DMIC1", "DMIC2", +}; + +static const struct soc_enum sidetone1_enum = + SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text); + +static const struct snd_kcontrol_new sidetone1_mux = + SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum); + +static const struct soc_enum sidetone2_enum = + SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text); + +static const struct snd_kcontrol_new sidetone2_mux = + SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum); + +static const char *aif1dac_text[] = { + "AIF1DACDAT", "AIF3DACDAT", +}; + +static const struct soc_enum aif1dac_enum = + SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); + +static const struct snd_kcontrol_new aif1dac_mux = + SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum); + +static const char *aif2dac_text[] = { + "AIF2DACDAT", "AIF3DACDAT", +}; + +static const struct soc_enum aif2dac_enum = + SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text); + +static const struct snd_kcontrol_new aif2dac_mux = + SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum); + +static const char *aif2adc_text[] = { + "AIF2ADCDAT", "AIF3DACDAT", +}; + +static const struct soc_enum aif2adc_enum = + SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text); + +static const struct snd_kcontrol_new aif2adc_mux = + SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum); + +static const char *aif3adc_text[] = { + "AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", +}; + +static const struct soc_enum aif3adc_enum = + SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text); + +static const struct snd_kcontrol_new aif3adc_mux = + SOC_DAPM_ENUM("AIF3ADC Mux", aif3adc_enum); + +static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("DMIC1DAT"), +SND_SOC_DAPM_INPUT("DMIC2DAT"), + +SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", + 0, WM8994_POWER_MANAGEMENT_4, 9, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", + 0, WM8994_POWER_MANAGEMENT_4, 8, 0), +SND_SOC_DAPM_AIF_IN("AIF1DAC1L", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 9, 0), +SND_SOC_DAPM_AIF_IN("AIF1DAC1R", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 8, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", + 0, WM8994_POWER_MANAGEMENT_4, 11, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", + 0, WM8994_POWER_MANAGEMENT_4, 10, 0), +SND_SOC_DAPM_AIF_IN("AIF1DAC2L", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 11, 0), +SND_SOC_DAPM_AIF_IN("AIF1DAC2R", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 10, 0), + +SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, + aif1adc1l_mix, ARRAY_SIZE(aif1adc1l_mix)), +SND_SOC_DAPM_MIXER("AIF1ADC1R Mixer", SND_SOC_NOPM, 0, 0, + aif1adc1r_mix, ARRAY_SIZE(aif1adc1r_mix)), + +SND_SOC_DAPM_MIXER("AIF2DAC2L Mixer", SND_SOC_NOPM, 0, 0, + aif2dac2l_mix, ARRAY_SIZE(aif2dac2l_mix)), +SND_SOC_DAPM_MIXER("AIF2DAC2R Mixer", SND_SOC_NOPM, 0, 0, + aif2dac2r_mix, ARRAY_SIZE(aif2dac2r_mix)), + +SND_SOC_DAPM_MUX("Left Sidetone", SND_SOC_NOPM, 0, 0, &sidetone1_mux), +SND_SOC_DAPM_MUX("Right Sidetone", SND_SOC_NOPM, 0, 0, &sidetone2_mux), + +SND_SOC_DAPM_MIXER("DAC1L Mixer", SND_SOC_NOPM, 0, 0, + dac1l_mix, ARRAY_SIZE(dac1l_mix)), +SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0, + dac1r_mix, ARRAY_SIZE(dac1r_mix)), + +SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0, + WM8994_POWER_MANAGEMENT_4, 13, 0), +SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0, + WM8994_POWER_MANAGEMENT_4, 12, 0), +SND_SOC_DAPM_AIF_IN("AIF2DACL", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 13, 0), +SND_SOC_DAPM_AIF_IN("AIF2DACR", NULL, 0, + WM8994_POWER_MANAGEMENT_5, 12, 0), + +SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), +SND_SOC_DAPM_MUX("AIF2DAC Mux", SND_SOC_NOPM, 0, 0, &aif2dac_mux), +SND_SOC_DAPM_MUX("AIF2ADC Mux", SND_SOC_NOPM, 0, 0, &aif2adc_mux), +SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &aif3adc_mux), + +SND_SOC_DAPM_AIF_IN("AIF3DACDAT", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_SUPPLY("TOCLK", WM8994_CLOCKING_1, 4, 0, NULL, 0), + +SND_SOC_DAPM_ADC("DMIC2L", NULL, WM8994_POWER_MANAGEMENT_4, 5, 0), +SND_SOC_DAPM_ADC("DMIC2R", NULL, WM8994_POWER_MANAGEMENT_4, 4, 0), +SND_SOC_DAPM_ADC("DMIC1L", NULL, WM8994_POWER_MANAGEMENT_4, 3, 0), +SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), + +/* Power is done with the muxes since the ADC power also controls the + * downsampling chain, the chip will automatically manage the analogue + * specific portions. + */ +SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), +SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), + +SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0), +SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), +SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), +SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), + +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), + +SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), +SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), + +SND_SOC_DAPM_POST("Debug log", post_ev), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + { "CLK_SYS", NULL, "AIF1CLK", check_clk_sys }, + { "CLK_SYS", NULL, "AIF2CLK", check_clk_sys }, + + { "DSP1CLK", NULL, "CLK_SYS" }, + { "DSP2CLK", NULL, "CLK_SYS" }, + { "DSPINTCLK", NULL, "CLK_SYS" }, + + { "AIF1ADC1L", NULL, "AIF1CLK" }, + { "AIF1ADC1L", NULL, "DSP1CLK" }, + { "AIF1ADC1R", NULL, "AIF1CLK" }, + { "AIF1ADC1R", NULL, "DSP1CLK" }, + { "AIF1ADC1R", NULL, "DSPINTCLK" }, + + { "AIF1DAC1L", NULL, "AIF1CLK" }, + { "AIF1DAC1L", NULL, "DSP1CLK" }, + { "AIF1DAC1R", NULL, "AIF1CLK" }, + { "AIF1DAC1R", NULL, "DSP1CLK" }, + { "AIF1DAC1R", NULL, "DSPINTCLK" }, + + { "AIF1ADC2L", NULL, "AIF1CLK" }, + { "AIF1ADC2L", NULL, "DSP1CLK" }, + { "AIF1ADC2R", NULL, "AIF1CLK" }, + { "AIF1ADC2R", NULL, "DSP1CLK" }, + { "AIF1ADC2R", NULL, "DSPINTCLK" }, + + { "AIF1DAC2L", NULL, "AIF1CLK" }, + { "AIF1DAC2L", NULL, "DSP1CLK" }, + { "AIF1DAC2R", NULL, "AIF1CLK" }, + { "AIF1DAC2R", NULL, "DSP1CLK" }, + { "AIF1DAC2R", NULL, "DSPINTCLK" }, + + { "AIF2ADCL", NULL, "AIF2CLK" }, + { "AIF2ADCL", NULL, "DSP2CLK" }, + { "AIF2ADCR", NULL, "AIF2CLK" }, + { "AIF2ADCR", NULL, "DSP2CLK" }, + { "AIF2ADCR", NULL, "DSPINTCLK" }, + + { "AIF2DACL", NULL, "AIF2CLK" }, + { "AIF2DACL", NULL, "DSP2CLK" }, + { "AIF2DACR", NULL, "AIF2CLK" }, + { "AIF2DACR", NULL, "DSP2CLK" }, + { "AIF2DACR", NULL, "DSPINTCLK" }, + + { "DMIC1L", NULL, "DMIC1DAT" }, + { "DMIC1L", NULL, "CLK_SYS" }, + { "DMIC1R", NULL, "DMIC1DAT" }, + { "DMIC1R", NULL, "CLK_SYS" }, + { "DMIC2L", NULL, "DMIC2DAT" }, + { "DMIC2L", NULL, "CLK_SYS" }, + { "DMIC2R", NULL, "DMIC2DAT" }, + { "DMIC2R", NULL, "CLK_SYS" }, + + { "ADCL", NULL, "AIF1CLK" }, + { "ADCL", NULL, "DSP1CLK" }, + { "ADCL", NULL, "DSPINTCLK" }, + + { "ADCR", NULL, "AIF1CLK" }, + { "ADCR", NULL, "DSP1CLK" }, + { "ADCR", NULL, "DSPINTCLK" }, + + { "ADCL Mux", "ADC", "ADCL" }, + { "ADCL Mux", "DMIC", "DMIC1L" }, + { "ADCR Mux", "ADC", "ADCR" }, + { "ADCR Mux", "DMIC", "DMIC1R" }, + + { "DAC1L", NULL, "AIF1CLK" }, + { "DAC1L", NULL, "DSP1CLK" }, + { "DAC1L", NULL, "DSPINTCLK" }, + + { "DAC1R", NULL, "AIF1CLK" }, + { "DAC1R", NULL, "DSP1CLK" }, + { "DAC1R", NULL, "DSPINTCLK" }, + + { "DAC2L", NULL, "AIF2CLK" }, + { "DAC2L", NULL, "DSP2CLK" }, + { "DAC2L", NULL, "DSPINTCLK" }, + + { "DAC2R", NULL, "AIF2DACR" }, + { "DAC2R", NULL, "AIF2CLK" }, + { "DAC2R", NULL, "DSP2CLK" }, + { "DAC2R", NULL, "DSPINTCLK" }, + + { "TOCLK", NULL, "CLK_SYS" }, + + /* AIF1 outputs */ + { "AIF1ADC1L", NULL, "AIF1ADC1L Mixer" }, + { "AIF1ADC1L Mixer", "ADC/DMIC Switch", "ADCL Mux" }, + { "AIF1ADC1L Mixer", "AIF2 Switch", "AIF2DACL" }, + + { "AIF1ADC1R", NULL, "AIF1ADC1R Mixer" }, + { "AIF1ADC1R Mixer", "ADC/DMIC Switch", "ADCR Mux" }, + { "AIF1ADC1R Mixer", "AIF2 Switch", "AIF2DACR" }, + + /* Pin level routing for AIF3 */ + { "AIF1DAC1L", NULL, "AIF1DAC Mux" }, + { "AIF1DAC1R", NULL, "AIF1DAC Mux" }, + { "AIF1DAC2L", NULL, "AIF1DAC Mux" }, + { "AIF1DAC2R", NULL, "AIF1DAC Mux" }, + + { "AIF2DACL", NULL, "AIF2DAC Mux" }, + { "AIF2DACR", NULL, "AIF2DAC Mux" }, + + { "AIF1DAC Mux", "AIF1DACDAT", "AIF1DACDAT" }, + { "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, + { "AIF2DAC Mux", "AIF2DACDAT", "AIF2DACDAT" }, + { "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, + { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, + { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, + { "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" }, + + /* DAC1 inputs */ + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" }, + { "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, + { "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, + { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, + { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, + + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" }, + { "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, + { "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, + { "DAC1R Mixer", "Left Sidetone Switch", "Left Sidetone" }, + { "DAC1R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + + /* DAC2/AIF2 outputs */ + { "AIF2ADCL", NULL, "AIF2DAC2L Mixer" }, + { "DAC2L", NULL, "AIF2DAC2L Mixer" }, + { "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" }, + { "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" }, + { "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" }, + { "AIF2DAC2L Mixer", "Left Sidetone Switch", "Left Sidetone" }, + { "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, + + { "AIF2ADCR", NULL, "AIF2DAC2R Mixer" }, + { "DAC2R", NULL, "AIF2DAC2R Mixer" }, + { "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" }, + { "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" }, + { "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" }, + { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, + { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, + + /* AIF3 output */ + { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1L" }, + { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1R" }, + { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2L" }, + { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2R" }, + { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCL" }, + { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCR" }, + { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, + { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + + /* Sidetone */ + { "Left Sidetone", "ADC/DMIC1", "ADCL Mux" }, + { "Left Sidetone", "DMIC2", "DMIC2L" }, + { "Right Sidetone", "ADC/DMIC1", "ADCR Mux" }, + { "Right Sidetone", "DMIC2", "DMIC2R" }, + + /* Output stages */ + { "Left Output Mixer", "DAC Switch", "DAC1L" }, + { "Right Output Mixer", "DAC Switch", "DAC1R" }, + + { "SPKL", "DAC1 Switch", "DAC1L" }, + { "SPKL", "DAC2 Switch", "DAC2L" }, + + { "SPKR", "DAC1 Switch", "DAC1R" }, + { "SPKR", "DAC2 Switch", "DAC2R" }, + + { "Left Headphone Mux", "DAC", "DAC1L" }, + { "Right Headphone Mux", "DAC", "DAC1R" }, +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +struct fll_div { + u16 outdiv; + u16 n; + u16 k; + u16 clk_ref_div; + u16 fll_fratio; +}; + +static int wm8994_get_fll_config(struct fll_div *fll, + int freq_in, int freq_out) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + + pr_debug("FLL input=%dHz, output=%dHz\n", freq_in, freq_out); + + /* Scale the input frequency down to <= 13.5MHz */ + fll->clk_ref_div = 0; + while (freq_in > 13500000) { + fll->clk_ref_div++; + freq_in /= 2; + + if (fll->clk_ref_div > 3) + return -EINVAL; + } + pr_debug("CLK_REF_DIV=%d, Fref=%dHz\n", fll->clk_ref_div, freq_in); + + /* Scale the output to give 90MHz<=Fvco<=100MHz */ + fll->outdiv = 3; + while (freq_out * (fll->outdiv + 1) < 90000000) { + fll->outdiv++; + if (fll->outdiv > 63) + return -EINVAL; + } + freq_out *= fll->outdiv + 1; + pr_debug("OUTDIV=%d, Fvco=%dHz\n", fll->outdiv, freq_out); + + if (freq_in > 1000000) { + fll->fll_fratio = 0; + } else { + fll->fll_fratio = 3; + freq_in *= 8; + } + pr_debug("FLL_FRATIO=%d, Fref=%dHz\n", fll->fll_fratio, freq_in); + + /* Now, calculate N.K */ + Ndiv = freq_out / freq_in; + + fll->n = Ndiv; + Nmod = freq_out % freq_in; + pr_debug("Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, freq_in); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + fll->k = K / 10; + + pr_debug("N=%x K=%x\n", fll->n, fll->k); + + return 0; +} + +static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8994_priv *wm8994 = codec->private_data; + int reg_offset, ret; + struct fll_div fll; + u16 reg, aif1, aif2; + + aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) + & WM8994_AIF1CLK_ENA; + + aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) + & WM8994_AIF2CLK_ENA; + + switch (id) { + case WM8994_FLL1: + reg_offset = 0; + id = 0; + break; + case WM8994_FLL2: + reg_offset = 0x20; + id = 1; + break; + default: + return -EINVAL; + } + + /* Are we changing anything? */ + if (wm8994->fll[id].src == src && + wm8994->fll[id].in == freq_in && wm8994->fll[id].out == freq_out) + return 0; + + /* If we're stopping the FLL redo the old config - no + * registers will actually be written but we avoid GCC flow + * analysis bugs spewing warnings. + */ + if (freq_out) + ret = wm8994_get_fll_config(&fll, freq_in, freq_out); + else + ret = wm8994_get_fll_config(&fll, wm8994->fll[id].in, + wm8994->fll[id].out); + if (ret < 0) + return ret; + + /* Gate the AIF clocks while we reclock */ + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA, 0); + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA, 0); + + /* We always need to disable the FLL while reconfiguring */ + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, + WM8994_FLL1_ENA, 0); + + reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) | + (fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset, + WM8994_FLL1_OUTDIV_MASK | + WM8994_FLL1_FRATIO_MASK, reg); + + snd_soc_write(codec, WM8994_FLL1_CONTROL_3 + reg_offset, fll.k); + + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_4 + reg_offset, + WM8994_FLL1_N_MASK, + fll.n << WM8994_FLL1_N_SHIFT); + + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8994_FLL1_REFCLK_DIV_MASK, + fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT); + + /* Enable (with fractional mode if required) */ + if (freq_out) { + if (fll.k) + reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC; + else + reg = WM8994_FLL1_ENA; + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, + WM8994_FLL1_ENA | WM8994_FLL1_FRAC, + reg); + } + + wm8994->fll[id].in = freq_in; + wm8994->fll[id].out = freq_out; + + /* Enable any gated AIF clocks */ + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA, aif1); + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA, aif2); + + configure_clock(codec); + + return 0; +} + +static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8994_priv *wm8994 = codec->private_data; + + switch (dai->id) { + case 1: + case 2: + break; + + default: + /* AIF3 shares clocking with AIF1/2 */ + return -EINVAL; + } + + switch (clk_id) { + case WM8994_SYSCLK_MCLK1: + wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_MCLK1; + wm8994->mclk[0] = freq; + dev_dbg(dai->dev, "AIF%d using MCLK1 at %uHz\n", + dai->id, freq); + break; + + case WM8994_SYSCLK_MCLK2: + /* TODO: Set GPIO AF */ + wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_MCLK2; + wm8994->mclk[1] = freq; + dev_dbg(dai->dev, "AIF%d using MCLK2 at %uHz\n", + dai->id, freq); + break; + + case WM8994_SYSCLK_FLL1: + wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_FLL1; + dev_dbg(dai->dev, "AIF%d using FLL1\n", dai->id); + break; + + case WM8994_SYSCLK_FLL2: + wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_FLL2; + dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id); + break; + + default: + return -EINVAL; + } + + configure_clock(codec); + + return 0; +} + +static int wm8994_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_VMID_SEL_MASK, 0x2); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Tweak DC servo configuration for improved + * performance. */ + snd_soc_write(codec, 0x102, 0x3); + snd_soc_write(codec, 0x56, 0x3); + snd_soc_write(codec, 0x102, 0); + + /* Discharge LINEOUT1 & 2 */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); + + /* Startup bias, VMID ramp & buffer */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (0x11 << WM8994_VMID_RAMP_SHIFT)); + + /* Main bias enable, VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, + WM8994_BIAS_ENA | 0x2); + + msleep(20); + } + + /* VMID=2x500k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_VMID_SEL_MASK, 0x4); + + break; + + case SND_SOC_BIAS_OFF: + /* Switch over to startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (1 << WM8994_VMID_RAMP_SHIFT)); + + /* Disable main biases */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0); + + /* Discharge line */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); + + msleep(5); + + /* Switch off startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, 0); + + break; + } + codec->bias_level = level; + return 0; +} + +static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int ms_reg; + int aif1_reg; + int ms = 0; + int aif1 = 0; + + switch (dai->id) { + case 1: + ms_reg = WM8994_AIF1_MASTER_SLAVE; + aif1_reg = WM8994_AIF1_CONTROL_1; + break; + case 2: + ms_reg = WM8994_AIF2_MASTER_SLAVE; + aif1_reg = WM8994_AIF2_CONTROL_1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + ms = WM8994_AIF1_MSTR; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif1 |= WM8994_AIF1_LRCLK_INV; + case SND_SOC_DAIFMT_DSP_A: + aif1 |= 0x18; + break; + case SND_SOC_DAIFMT_I2S: + aif1 |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif1 |= 0x8; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8994_AIF1_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8994_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif1 |= WM8994_AIF1_LRCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, aif1_reg, + WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV | + WM8994_AIF1_FMT_MASK, + aif1); + snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, + ms); + + return 0; +} + +static struct { + int val, rate; +} srs[] = { + { 0, 8000 }, + { 1, 11025 }, + { 2, 12000 }, + { 3, 16000 }, + { 4, 22050 }, + { 5, 24000 }, + { 6, 32000 }, + { 7, 44100 }, + { 8, 48000 }, + { 9, 88200 }, + { 10, 96000 }, +}; + +static int fs_ratios[] = { + 64, 128, 192, 256, 348, 512, 768, 1024, 1408, 1536 +}; + +static int bclk_divs[] = { + 10, 15, 20, 30, 40, 50, 60, 80, 110, 120, 160, 220, 240, 320, 440, 480, + 640, 880, 960, 1280, 1760, 1920 +}; + +static int wm8994_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8994_priv *wm8994 = codec->private_data; + int aif1_reg; + int bclk_reg; + int lrclk_reg; + int rate_reg; + int aif1 = 0; + int bclk = 0; + int lrclk = 0; + int rate_val = 0; + int id = dai->id - 1; + + int i, cur_val, best_val, bclk_rate, best; + + switch (dai->id) { + case 1: + aif1_reg = WM8994_AIF1_CONTROL_1; + bclk_reg = WM8994_AIF1_BCLK; + rate_reg = WM8994_AIF1_RATE; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + wm8994->lrclk_shared[0]) + lrclk_reg = WM8994_AIF1DAC_LRCLK; + else + lrclk_reg = WM8994_AIF1ADC_LRCLK; + break; + case 2: + aif1_reg = WM8994_AIF2_CONTROL_1; + bclk_reg = WM8994_AIF2_BCLK; + rate_reg = WM8994_AIF2_RATE; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + wm8994->lrclk_shared[1]) + lrclk_reg = WM8994_AIF2DAC_LRCLK; + else + lrclk_reg = WM8994_AIF2ADC_LRCLK; + break; + default: + return -EINVAL; + } + + bclk_rate = params_rate(params) * 2; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + bclk_rate *= 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + bclk_rate *= 20; + aif1 |= 0x20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bclk_rate *= 24; + aif1 |= 0x40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bclk_rate *= 32; + aif1 |= 0x60; + break; + default: + return -EINVAL; + } + + /* Try to find an appropriate sample rate; look for an exact match. */ + for (i = 0; i < ARRAY_SIZE(srs); i++) + if (srs[i].rate == params_rate(params)) + break; + if (i == ARRAY_SIZE(srs)) + return -EINVAL; + rate_val |= srs[i].val << WM8994_AIF1_SR_SHIFT; + + dev_dbg(dai->dev, "Sample rate is %dHz\n", srs[i].rate); + dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n", + dai->id, wm8994->aifclk[id], bclk_rate); + + if (wm8994->aifclk[id] == 0) { + dev_err(dai->dev, "AIF%dCLK not configured\n", dai->id); + return -EINVAL; + } + + /* AIFCLK/fs ratio; look for a close match in either direction */ + best = 0; + best_val = abs((fs_ratios[0] * params_rate(params)) + - wm8994->aifclk[id]); + for (i = 1; i < ARRAY_SIZE(fs_ratios); i++) { + cur_val = abs((fs_ratios[i] * params_rate(params)) + - wm8994->aifclk[id]); + if (cur_val >= best_val) + continue; + best = i; + best_val = cur_val; + } + dev_dbg(dai->dev, "Selected AIF%dCLK/fs = %d\n", + dai->id, fs_ratios[best]); + rate_val |= best; + + /* We may not get quite the right frequency if using + * approximate clocks so look for the closest match that is + * higher than the target (we need to ensure that there enough + * BCLKs to clock out the samples). + */ + best = 0; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + if (bclk_divs[i] < 0) + continue; + cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) + - bclk_rate * 10; + if (cur_val < 0) /* BCLK table is sorted */ + break; + best = i; + } + bclk_rate = wm8994->aifclk[id] / bclk_divs[best]; + dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", + bclk_divs[best], bclk_rate); + bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; + + lrclk = bclk_rate / params_rate(params); + dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", + lrclk, bclk_rate / lrclk); + + snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); + snd_soc_update_bits(codec, bclk_reg, WM8994_AIF1_BCLK_DIV_MASK, bclk); + snd_soc_update_bits(codec, lrclk_reg, WM8994_AIF1DAC_RATE_MASK, + lrclk); + snd_soc_update_bits(codec, rate_reg, WM8994_AIF1_SR_MASK | + WM8994_AIF1CLK_RATE_MASK, rate_val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (dai->id) { + case 1: + wm8994->dac_rates[0] = params_rate(params); + wm8994_set_retune_mobile(codec, 0); + wm8994_set_retune_mobile(codec, 1); + break; + case 2: + wm8994->dac_rates[1] = params_rate(params); + wm8994_set_retune_mobile(codec, 2); + break; + } + } + + return 0; +} + +static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int mute_reg; + int reg; + + switch (codec_dai->id) { + case 1: + mute_reg = WM8994_AIF1_DAC1_FILTERS_1; + break; + case 2: + mute_reg = WM8994_AIF2_DAC_FILTERS_1; + break; + default: + return -EINVAL; + } + + if (mute) + reg = WM8994_AIF1DAC1_MUTE; + else + reg = 0; + + snd_soc_update_bits(codec, mute_reg, WM8994_AIF1DAC1_MUTE, reg); + + return 0; +} + +#define WM8994_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { + .set_sysclk = wm8994_set_dai_sysclk, + .set_fmt = wm8994_set_dai_fmt, + .hw_params = wm8994_hw_params, + .digital_mute = wm8994_aif_mute, + .set_pll = wm8994_set_fll, +}; + +static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { + .set_sysclk = wm8994_set_dai_sysclk, + .set_fmt = wm8994_set_dai_fmt, + .hw_params = wm8994_hw_params, + .digital_mute = wm8994_aif_mute, + .set_pll = wm8994_set_fll, +}; + +struct snd_soc_dai wm8994_dai[] = { + { + .name = "WM8994 AIF1", + .id = 1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .ops = &wm8994_aif1_dai_ops, + }, + { + .name = "WM8994 AIF2", + .id = 2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .ops = &wm8994_aif2_dai_ops, + }, + { + .name = "WM8994 AIF3", + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + .playback = { + .stream_name = "AIF3 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8994_RATES, + .formats = WM8994_FORMATS, + }, + } +}; +EXPORT_SYMBOL_GPL(wm8994_dai); + +#ifdef CONFIG_PM +static int wm8994_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8994_priv *wm8994 = codec->private_data; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { + memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], + sizeof(struct fll_config)); + ret = wm8994_set_fll(&codec->dai[0], i + 1, 0, 0, 0); + if (ret < 0) + dev_warn(codec->dev, "Failed to stop FLL%d: %d\n", + i + 1, ret); + } + + wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8994_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8994_priv *wm8994 = codec->private_data; + u16 *reg_cache = codec->reg_cache; + int i, ret; + + /* Restore the registers */ + for (i = 1; i < ARRAY_SIZE(wm8994->reg_cache); i++) { + switch (i) { + case WM8994_LDO_1: + case WM8994_LDO_2: + case WM8994_SOFTWARE_RESET: + /* Handled by other MFD drivers */ + continue; + default: + break; + } + + if (!access_masks[i].writable) + continue; + + wm8994_reg_write(codec->control_data, i, reg_cache[i]); + } + + wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { + ret = wm8994_set_fll(&codec->dai[0], i + 1, + wm8994->fll_suspend[i].src, + wm8994->fll_suspend[i].in, + wm8994->fll_suspend[i].out); + if (ret < 0) + dev_warn(codec->dev, "Failed to restore FLL%d: %d\n", + i + 1, ret); + } + + return 0; +} +#else +#define wm8994_suspend NULL +#define wm8994_resume NULL +#endif + +static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) +{ + struct snd_soc_codec *codec = &wm8994->codec; + struct wm8994_pdata *pdata = wm8994->pdata; + struct snd_kcontrol_new controls[] = { + SOC_ENUM_EXT("AIF1.1 EQ Mode", + wm8994->retune_mobile_enum, + wm8994_get_retune_mobile_enum, + wm8994_put_retune_mobile_enum), + SOC_ENUM_EXT("AIF1.2 EQ Mode", + wm8994->retune_mobile_enum, + wm8994_get_retune_mobile_enum, + wm8994_put_retune_mobile_enum), + SOC_ENUM_EXT("AIF2 EQ Mode", + wm8994->retune_mobile_enum, + wm8994_get_retune_mobile_enum, + wm8994_put_retune_mobile_enum), + }; + int ret, i, j; + const char **t; + + /* We need an array of texts for the enum API but the number + * of texts is likely to be less than the number of + * configurations due to the sample rate dependency of the + * configurations. */ + wm8994->num_retune_mobile_texts = 0; + wm8994->retune_mobile_texts = NULL; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + for (j = 0; j < wm8994->num_retune_mobile_texts; j++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8994->retune_mobile_texts[j]) == 0) + break; + } + + if (j != wm8994->num_retune_mobile_texts) + continue; + + /* Expand the array... */ + t = krealloc(wm8994->retune_mobile_texts, + sizeof(char *) * + (wm8994->num_retune_mobile_texts + 1), + GFP_KERNEL); + if (t == NULL) + continue; + + /* ...store the new entry... */ + t[wm8994->num_retune_mobile_texts] = + pdata->retune_mobile_cfgs[i].name; + + /* ...and remember the new version. */ + wm8994->num_retune_mobile_texts++; + wm8994->retune_mobile_texts = t; + } + + dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", + wm8994->num_retune_mobile_texts); + + wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts; + wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts; + + ret = snd_soc_add_controls(&wm8994->codec, controls, + ARRAY_SIZE(controls)); + if (ret != 0) + dev_err(wm8994->codec.dev, + "Failed to add ReTune Mobile controls: %d\n", ret); +} + +static void wm8994_handle_pdata(struct wm8994_priv *wm8994) +{ + struct snd_soc_codec *codec = &wm8994->codec; + struct wm8994_pdata *pdata = wm8994->pdata; + int ret, i; + + if (!pdata) + return; + + wm_hubs_handle_analogue_pdata(codec, pdata->lineout1_diff, + pdata->lineout2_diff, + pdata->lineout1fb, + pdata->lineout2fb, + pdata->jd_scthr, + pdata->jd_thr, + pdata->micbias1_lvl, + pdata->micbias2_lvl); + + dev_dbg(codec->dev, "%d DRC configurations\n", pdata->num_drc_cfgs); + + if (pdata->num_drc_cfgs) { + struct snd_kcontrol_new controls[] = { + SOC_ENUM_EXT("AIF1DRC1 Mode", wm8994->drc_enum, + wm8994_get_drc_enum, wm8994_put_drc_enum), + SOC_ENUM_EXT("AIF1DRC2 Mode", wm8994->drc_enum, + wm8994_get_drc_enum, wm8994_put_drc_enum), + SOC_ENUM_EXT("AIF2DRC Mode", wm8994->drc_enum, + wm8994_get_drc_enum, wm8994_put_drc_enum), + }; + + /* We need an array of texts for the enum API */ + wm8994->drc_texts = kmalloc(sizeof(char *) + * pdata->num_drc_cfgs, GFP_KERNEL); + if (!wm8994->drc_texts) { + dev_err(wm8994->codec.dev, + "Failed to allocate %d DRC config texts\n", + pdata->num_drc_cfgs); + return; + } + + for (i = 0; i < pdata->num_drc_cfgs; i++) + wm8994->drc_texts[i] = pdata->drc_cfgs[i].name; + + wm8994->drc_enum.max = pdata->num_drc_cfgs; + wm8994->drc_enum.texts = wm8994->drc_texts; + + ret = snd_soc_add_controls(&wm8994->codec, controls, + ARRAY_SIZE(controls)); + if (ret != 0) + dev_err(wm8994->codec.dev, + "Failed to add DRC mode controls: %d\n", ret); + + for (i = 0; i < WM8994_NUM_DRC; i++) + wm8994_set_drc(codec, i); + } + + dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", + pdata->num_retune_mobile_cfgs); + + if (pdata->num_retune_mobile_cfgs) + wm8994_handle_retune_mobile_pdata(wm8994); + else + snd_soc_add_controls(&wm8994->codec, wm8994_eq_controls, + ARRAY_SIZE(wm8994_eq_controls)); +} + +static int wm8994_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8994_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8994_codec; + codec = wm8994_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + return ret; + } + + wm8994_handle_pdata(codec->private_data); + + wm_hubs_add_analogue_controls(codec); + snd_soc_add_controls(codec, wm8994_snd_controls, + ARRAY_SIZE(wm8994_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8994_dapm_widgets, + ARRAY_SIZE(wm8994_dapm_widgets)); + wm_hubs_add_analogue_routes(codec, 0, 0); + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + return 0; +} + +static int wm8994_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8994 = { + .probe = wm8994_probe, + .remove = wm8994_remove, + .suspend = wm8994_suspend, + .resume = wm8994_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8994); + +static int wm8994_codec_probe(struct platform_device *pdev) +{ + int ret; + struct wm8994_priv *wm8994; + struct snd_soc_codec *codec; + int i; + u16 rev; + + if (wm8994_codec) { + dev_err(&pdev->dev, "Another WM8994 is registered\n"); + return -EINVAL; + } + + wm8994 = kzalloc(sizeof(struct wm8994_priv), GFP_KERNEL); + if (!wm8994) { + dev_err(&pdev->dev, "Failed to allocate private data\n"); + return -ENOMEM; + } + + codec = &wm8994->codec; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8994; + codec->control_data = dev_get_drvdata(pdev->dev.parent); + codec->name = "WM8994"; + codec->owner = THIS_MODULE; + codec->read = wm8994_read; + codec->write = wm8994_write; + codec->readable_register = wm8994_readable; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8994_set_bias_level; + codec->dai = &wm8994_dai[0]; + codec->num_dai = 3; + codec->reg_cache_size = WM8994_MAX_REGISTER; + codec->reg_cache = &wm8994->reg_cache; + codec->dev = &pdev->dev; + + wm8994->pdata = pdev->dev.parent->platform_data; + + /* Fill the cache with physical values we inherited; don't reset */ + ret = wm8994_bulk_read(codec->control_data, 0, + ARRAY_SIZE(wm8994->reg_cache) - 1, + codec->reg_cache); + if (ret < 0) { + dev_err(codec->dev, "Failed to fill register cache: %d\n", + ret); + goto err; + } + + /* Clear the cached values for unreadable/volatile registers to + * avoid potential confusion. + */ + for (i = 0; i < ARRAY_SIZE(wm8994->reg_cache); i++) + if (wm8994_volatile(i) || !wm8994_readable(i)) + wm8994->reg_cache[i] = 0; + + /* Set revision-specific configuration */ + rev = snd_soc_read(codec, WM8994_CHIP_REVISION); + switch (rev) { + case 2: + case 3: + wm8994->hubs.dcs_codes = -5; + wm8994->hubs.hp_startup_mode = 1; + break; + default: + break; + } + + + /* Remember if AIFnLRCLK is configured as a GPIO. This should be + * configured on init - if a system wants to do this dynamically + * at runtime we can deal with that then. + */ + ret = wm8994_reg_read(codec->control_data, WM8994_GPIO_1); + if (ret < 0) { + dev_err(codec->dev, "Failed to read GPIO1 state: %d\n", ret); + goto err; + } + if ((ret & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) { + wm8994->lrclk_shared[0] = 1; + wm8994_dai[0].symmetric_rates = 1; + } else { + wm8994->lrclk_shared[0] = 0; + } + + ret = wm8994_reg_read(codec->control_data, WM8994_GPIO_6); + if (ret < 0) { + dev_err(codec->dev, "Failed to read GPIO6 state: %d\n", ret); + goto err; + } + if ((ret & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) { + wm8994->lrclk_shared[1] = 1; + wm8994_dai[1].symmetric_rates = 1; + } else { + wm8994->lrclk_shared[1] = 0; + } + + for (i = 0; i < ARRAY_SIZE(wm8994_dai); i++) + wm8994_dai[i].dev = codec->dev; + + wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8994_codec = codec; + + /* Latch volume updates (right only; we always do left then right). */ + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME, + WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME, + WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME, + WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME, + WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME, + WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME, + WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME, + WM8994_DAC1_VU, WM8994_DAC1_VU); + snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME, + WM8994_DAC2_VU, WM8994_DAC2_VU); + + /* Set the low bit of the 3D stereo depth so TLV matches */ + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_FILTERS_2, + 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, + 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_FILTERS_2, + 1 << WM8994_AIF1DAC2_3D_GAIN_SHIFT, + 1 << WM8994_AIF1DAC2_3D_GAIN_SHIFT); + snd_soc_update_bits(codec, WM8994_AIF2_DAC_FILTERS_2, + 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, + 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); + + wm8994_update_class_w(codec); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(wm8994_dai, ARRAY_SIZE(wm8994_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + platform_set_drvdata(pdev, wm8994); + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8994); + return ret; +} + +static int __devexit wm8994_codec_remove(struct platform_device *pdev) +{ + struct wm8994_priv *wm8994 = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = &wm8994->codec; + + wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(wm8994_dai, ARRAY_SIZE(wm8994_dai)); + snd_soc_unregister_codec(&wm8994->codec); + kfree(wm8994); + wm8994_codec = NULL; + + return 0; +} + +static struct platform_driver wm8994_codec_driver = { + .driver = { + .name = "wm8994-codec", + .owner = THIS_MODULE, + }, + .probe = wm8994_codec_probe, + .remove = __devexit_p(wm8994_codec_remove), +}; + +static __init int wm8994_init(void) +{ + return platform_driver_register(&wm8994_codec_driver); +} +module_init(wm8994_init); + +static __exit void wm8994_exit(void) +{ + platform_driver_unregister(&wm8994_codec_driver); +} +module_exit(wm8994_exit); + + +MODULE_DESCRIPTION("ASoC WM8994 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8994-codec"); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h new file mode 100644 index 0000000..0a5e142 --- /dev/null +++ b/sound/soc/codecs/wm8994.h @@ -0,0 +1,26 @@ +/* + * wm8994.h -- WM8994 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8994_H +#define _WM8994_H + +#include + +extern struct snd_soc_codec_device soc_codec_dev_wm8994; +extern struct snd_soc_dai wm8994_dai[]; + +/* Sources for AIF1/2 SYSCLK - use with set_dai_sysclk() */ +#define WM8994_SYSCLK_MCLK1 1 +#define WM8994_SYSCLK_MCLK2 2 +#define WM8994_SYSCLK_FLL1 3 +#define WM8994_SYSCLK_FLL2 4 + +#define WM8994_FLL1 1 +#define WM8994_FLL2 2 + +#endif -- cgit v0.10.2 From c85a400499093b2025238413198e48e4d825723e Mon Sep 17 00:00:00 2001 From: Thadeu Lima de Souza Cascardo Date: Mon, 1 Feb 2010 16:17:01 -0200 Subject: ALSA: trivial: sound seq ioctl dbg: print hexadecimal value padded with 0s Instead of padding with blanks and printing "number=0x a", print "number=0x0a". Signed-off-by: Thadeu Lima de Souza Cascardo Signed-off-by: Takashi Iwai diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 8ca2be3..48eca9f 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2190,7 +2190,7 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, if (p->cmd == cmd) return p->func(client, arg); } - snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%2x)\n", + snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%02x)\n", cmd, _IOC_TYPE(cmd), _IOC_NR(cmd)); return -ENOTTY; } -- cgit v0.10.2 From fead215d1c0a385fc27a1fa96b7abbc4d66fb4c6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Feb 2010 10:06:55 +0000 Subject: ASoC: Fix WM8994 dependency The dependency on MFD_WM8994 rather than I2C went awry. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6b8a101..5ab5921 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C - select SND_SOC_WM8994 if I2C + select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM9081 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS -- cgit v0.10.2 From 07cd8ada1aba5556b0d5d2264ce0f40d1ff1d131 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Tue, 2 Feb 2010 18:53:19 +0900 Subject: ASoC: Fix BCLK calculation of WM8994 This fixes BCLK calculation and removes unnecessary check code. Signed-off-by: Joonyoung Shim Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5dd4b29..29f3771 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3267,15 +3267,12 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, */ best = 0; for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { - if (bclk_divs[i] < 0) - continue; - cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - - bclk_rate * 10; + cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - bclk_rate; if (cur_val < 0) /* BCLK table is sorted */ break; best = i; } - bclk_rate = wm8994->aifclk[id] / bclk_divs[best]; + bclk_rate = wm8994->aifclk[id] * 10 / bclk_divs[best]; dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", bclk_divs[best], bclk_rate); bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; -- cgit v0.10.2 From 59cdd9bc057a54384a7838231dd2672a89dff2ac Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Mon, 1 Feb 2010 23:22:16 -0800 Subject: ASoC: Fix continuation line formats String constants that are continued on subsequent lines with \ are not good. Signed-off-by: Joe Perches Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c index 9e61a7c..a98f40c 100644 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -229,8 +229,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, spin_unlock_irqrestore(&pcm->lock, flags); - dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs \ - SCLK_DIV=%d SYNC_DIV=%d\n", + dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs SCLK_DIV=%d SYNC_DIV=%d\n", clk_get_rate(clk), pcm->sclk_per_fs, sclk_div, sync_div); -- cgit v0.10.2 From 026384d614b827f368834860c9b623ce08439b7e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 2 Feb 2010 18:45:27 +0800 Subject: ASoC: fix PXA SSP port resume Unconditionally save the register states when suspending and restore them again at resume time. Register contents were not preserved over suspend, and hence the driver takes false assumptions about them. The clock must be enabled to access the register block. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3bd7712..e69397f 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -135,10 +135,11 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_priv *priv = cpu_dai->private_data; if (!cpu_dai->active) - return 0; + clk_enable(priv->dev.ssp->clk); ssp_save_state(&priv->dev, &priv->state); clk_disable(priv->dev.ssp->clk); + return 0; } @@ -146,12 +147,13 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) { struct ssp_priv *priv = cpu_dai->private_data; - if (!cpu_dai->active) - return 0; - clk_enable(priv->dev.ssp->clk); ssp_restore_state(&priv->dev, &priv->state); - ssp_enable(&priv->dev); + + if (cpu_dai->active) + ssp_enable(&priv->dev); + else + clk_disable(priv->dev.ssp->clk); return 0; } -- cgit v0.10.2 From d5e1ca05f758fec2845a97fd7aa1eeca91c51a21 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 2 Feb 2010 17:48:51 +0100 Subject: ALSA: dummy driver - add model parameter This is a cleanup for the dummy driver. The model kernel module parameter is introduced to select the soundcard emulation. Signed-off-by: Jaroslav Kysela diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 252e04c..7f41990 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -45,109 +45,23 @@ MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}"); #define MAX_PCM_SUBSTREAMS 128 #define MAX_MIDI_DEVICES 2 -#if 0 /* emu10k1 emulation */ -#define MAX_BUFFER_SIZE (128 * 1024) -static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) -{ - int err; - err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (err < 0) - return err; - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); - if (err < 0) - return err; - return 0; -} -#define add_playback_constraints emu10k1_playback_constraints -#endif - -#if 0 /* RME9652 emulation */ -#define MAX_BUFFER_SIZE (26 * 64 * 1024) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S32_LE -#define USE_CHANNELS_MIN 26 -#define USE_CHANNELS_MAX 26 -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 2 -#endif - -#if 0 /* ICE1712 emulation */ -#define MAX_BUFFER_SIZE (256 * 1024) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S32_LE -#define USE_CHANNELS_MIN 10 -#define USE_CHANNELS_MAX 10 -#define USE_PERIODS_MIN 1 -#define USE_PERIODS_MAX 1024 -#endif - -#if 0 /* UDA1341 emulation */ -#define MAX_BUFFER_SIZE (16380) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 255 -#endif - -#if 0 /* simple AC97 bridge (intel8x0) with 48kHz AC97 only codec */ -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_RATE SNDRV_PCM_RATE_48000 -#define USE_RATE_MIN 48000 -#define USE_RATE_MAX 48000 -#endif - -#if 0 /* CA0106 */ -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_RATE (SNDRV_PCM_RATE_48000|SNDRV_PCM_RATE_96000|SNDRV_PCM_RATE_192000) -#define USE_RATE_MIN 48000 -#define USE_RATE_MAX 192000 -#define MAX_BUFFER_SIZE ((65536-64)*8) -#define MAX_PERIOD_SIZE (65536-64) -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 8 -#endif - - /* defaults */ -#ifndef MAX_BUFFER_SIZE #define MAX_BUFFER_SIZE (64*1024) -#endif -#ifndef MAX_PERIOD_SIZE +#define MIN_PERIOD_SIZE 64 #define MAX_PERIOD_SIZE MAX_BUFFER_SIZE -#endif -#ifndef USE_FORMATS #define USE_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) -#endif -#ifndef USE_RATE #define USE_RATE SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000 #define USE_RATE_MIN 5500 #define USE_RATE_MAX 48000 -#endif -#ifndef USE_CHANNELS_MIN #define USE_CHANNELS_MIN 1 -#endif -#ifndef USE_CHANNELS_MAX #define USE_CHANNELS_MAX 2 -#endif -#ifndef USE_PERIODS_MIN #define USE_PERIODS_MIN 1 -#endif -#ifndef USE_PERIODS_MAX #define USE_PERIODS_MAX 1024 -#endif -#ifndef add_playback_constraints -#define add_playback_constraints(x) 0 -#endif -#ifndef add_capture_constraints -#define add_capture_constraints(x) 0 -#endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; +static char *model[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = NULL}; static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; //static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; @@ -162,6 +76,8 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for dummy soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable this dummy soundcard."); +module_param_array(model, charp, NULL, 0444); +MODULE_PARM_DESC(model, "Soundcard model."); module_param_array(pcm_devs, int, NULL, 0444); MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver."); module_param_array(pcm_substreams, int, NULL, 0444); @@ -193,9 +109,28 @@ struct dummy_timer_ops { snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *); }; +struct dummy_model { + const char *name; + int (*playback_constraints)(struct snd_pcm_runtime *runtime); + int (*capture_constraints)(struct snd_pcm_runtime *runtime); + u64 formats; + size_t buffer_bytes_max; + size_t period_bytes_min; + size_t period_bytes_max; + unsigned int periods_min; + unsigned int periods_max; + unsigned int rates; + unsigned int rate_min; + unsigned int rate_max; + unsigned int channels_min; + unsigned int channels_max; +}; + struct snd_dummy { struct snd_card *card; + struct dummy_model *model; struct snd_pcm *pcm; + struct snd_pcm_hardware pcm_hw; spinlock_t mixer_lock; int mixer_volume[MIXER_ADDR_LAST+1][2]; int capture_source[MIXER_ADDR_LAST+1][2]; @@ -203,6 +138,92 @@ struct snd_dummy { }; /* + * card models + */ + +static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) +{ + int err; + err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) + return err; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); + if (err < 0) + return err; + return 0; +} + +struct dummy_model model_emu10k1 = { + .name = "emu10k1", + .playback_constraints = emu10k1_playback_constraints, + .buffer_bytes_max = 128 * 1024, +}; + +struct dummy_model model_rme9652 = { + .name = "rme9652", + .buffer_bytes_max = 26 * 64 * 1024, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 26, + .channels_max = 26, + .periods_min = 2, + .periods_max = 2, +}; + +struct dummy_model model_ice1712 = { + .name = "ice1712", + .buffer_bytes_max = 256 * 1024, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 10, + .channels_max = 10, + .periods_min = 1, + .periods_max = 1024, +}; + +struct dummy_model model_uda1341 = { + .name = "uda1341", + .buffer_bytes_max = 16380, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 2, + .periods_min = 2, + .periods_max = 255, +}; + +struct dummy_model model_ac97 = { + .name = "ac97", + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, +}; + +struct dummy_model model_ca0106 = { + .name = "ca0106", + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .buffer_bytes_max = ((65536-64)*8), + .period_bytes_max = (65536-64), + .periods_min = 2, + .periods_max = 8, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000|SNDRV_PCM_RATE_96000|SNDRV_PCM_RATE_192000, + .rate_min = 48000, + .rate_max = 192000, +}; + +struct dummy_model *dummy_models[] = { + &model_emu10k1, + &model_rme9652, + &model_ice1712, + &model_uda1341, + &model_ac97, + &model_ca0106, + NULL +}; + +/* * system timer interface */ @@ -509,7 +530,7 @@ static struct snd_pcm_hardware dummy_pcm_hardware = { .channels_min = USE_CHANNELS_MIN, .channels_max = USE_CHANNELS_MAX, .buffer_bytes_max = MAX_BUFFER_SIZE, - .period_bytes_min = 64, + .period_bytes_min = MIN_PERIOD_SIZE, .period_bytes_max = MAX_PERIOD_SIZE, .periods_min = USE_PERIODS_MIN, .periods_max = USE_PERIODS_MAX, @@ -538,6 +559,7 @@ static int dummy_pcm_hw_free(struct snd_pcm_substream *substream) static int dummy_pcm_open(struct snd_pcm_substream *substream) { struct snd_dummy *dummy = snd_pcm_substream_chip(substream); + struct dummy_model *model = dummy->model; struct snd_pcm_runtime *runtime = substream->runtime; int err; @@ -551,7 +573,7 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) if (err < 0) return err; - runtime->hw = dummy_pcm_hardware; + runtime->hw = dummy->pcm_hw; if (substream->pcm->device & 1) { runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED; runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED; @@ -560,10 +582,16 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = add_playback_constraints(substream->runtime); - else - err = add_capture_constraints(substream->runtime); + if (model == NULL) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (model->playback_constraints) + err = model->playback_constraints(substream->runtime); + } else { + if (model->capture_constraints) + err = model->capture_constraints(substream->runtime); + } if (err < 0) { dummy->timer_ops->free(substream); return err; @@ -823,17 +851,19 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy) /* * proc interface */ -static void print_formats(struct snd_info_buffer *buffer) +static void print_formats(struct snd_dummy *dummy, + struct snd_info_buffer *buffer) { int i; for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { - if (dummy_pcm_hardware.formats & (1ULL << i)) + if (dummy->pcm_hw.formats & (1ULL << i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } } -static void print_rates(struct snd_info_buffer *buffer) +static void print_rates(struct snd_dummy *dummy, + struct snd_info_buffer *buffer) { static int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, @@ -841,19 +871,19 @@ static void print_rates(struct snd_info_buffer *buffer) }; int i; - if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_CONTINUOUS) + if (dummy->pcm_hw.rates & SNDRV_PCM_RATE_CONTINUOUS) snd_iprintf(buffer, " continuous"); - if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_KNOT) + if (dummy->pcm_hw.rates & SNDRV_PCM_RATE_KNOT) snd_iprintf(buffer, " knot"); for (i = 0; i < ARRAY_SIZE(rates); i++) - if (dummy_pcm_hardware.rates & (1 << i)) + if (dummy->pcm_hw.rates & (1 << i)) snd_iprintf(buffer, " %d", rates[i]); } -#define get_dummy_int_ptr(ofs) \ - (unsigned int *)((char *)&dummy_pcm_hardware + (ofs)) -#define get_dummy_ll_ptr(ofs) \ - (unsigned long long *)((char *)&dummy_pcm_hardware + (ofs)) +#define get_dummy_int_ptr(dummy, ofs) \ + (unsigned int *)((char *)&((dummy)->pcm_hw) + (ofs)) +#define get_dummy_ll_ptr(dummy, ofs) \ + (unsigned long long *)((char *)&((dummy)->pcm_hw) + (ofs)) struct dummy_hw_field { const char *name; @@ -884,20 +914,21 @@ static struct dummy_hw_field fields[] = { static void dummy_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { + struct snd_dummy *dummy = entry->private_data; int i; for (i = 0; i < ARRAY_SIZE(fields); i++) { snd_iprintf(buffer, "%s ", fields[i].name); if (fields[i].size == sizeof(int)) snd_iprintf(buffer, fields[i].format, - *get_dummy_int_ptr(fields[i].offset)); + *get_dummy_int_ptr(dummy, fields[i].offset)); else snd_iprintf(buffer, fields[i].format, - *get_dummy_ll_ptr(fields[i].offset)); + *get_dummy_ll_ptr(dummy, fields[i].offset)); if (!strcmp(fields[i].name, "formats")) - print_formats(buffer); + print_formats(dummy, buffer); else if (!strcmp(fields[i].name, "rates")) - print_rates(buffer); + print_rates(dummy, buffer); snd_iprintf(buffer, "\n"); } } @@ -905,6 +936,7 @@ static void dummy_proc_read(struct snd_info_entry *entry, static void dummy_proc_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { + struct snd_dummy *dummy = entry->private_data; char line[64]; while (!snd_info_get_line(buffer, line, sizeof(line))) { @@ -924,9 +956,9 @@ static void dummy_proc_write(struct snd_info_entry *entry, if (strict_strtoull(item, 0, &val)) continue; if (fields[i].size == sizeof(int)) - *get_dummy_int_ptr(fields[i].offset) = val; + *get_dummy_int_ptr(dummy, fields[i].offset) = val; else - *get_dummy_ll_ptr(fields[i].offset) = val; + *get_dummy_ll_ptr(dummy, fields[i].offset) = val; } } @@ -938,6 +970,7 @@ static void __devinit dummy_proc_init(struct snd_dummy *chip) snd_info_set_text_ops(entry, chip, dummy_proc_read); entry->c.text.write = dummy_proc_write; entry->mode |= S_IWUSR; + entry->private_data = chip; } } #else @@ -948,6 +981,7 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) { struct snd_card *card; struct snd_dummy *dummy; + struct dummy_model *m = NULL, **mdl; int idx, err; int dev = devptr->id; @@ -957,6 +991,15 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) return err; dummy = card->private_data; dummy->card = card; + for (mdl = dummy_models; *mdl && model[dev]; mdl++) { + if (strcmp(model[dev], (*mdl)->name) == 0) { + printk(KERN_INFO + "snd-dummy: Using model '%s' for card %i\n", + (*mdl)->name, card->number); + m = dummy->model = *mdl; + break; + } + } for (idx = 0; idx < MAX_PCM_DEVICES && idx < pcm_devs[dev]; idx++) { if (pcm_substreams[dev] < 1) pcm_substreams[dev] = 1; @@ -966,6 +1009,33 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) if (err < 0) goto __nodev; } + + dummy->pcm_hw = dummy_pcm_hardware; + if (m) { + if (m->formats) + dummy->pcm_hw.formats = m->formats; + if (m->buffer_bytes_max) + dummy->pcm_hw.buffer_bytes_max = m->buffer_bytes_max; + if (m->period_bytes_min) + dummy->pcm_hw.period_bytes_min = m->period_bytes_min; + if (m->period_bytes_max) + dummy->pcm_hw.period_bytes_max = m->period_bytes_max; + if (m->periods_min) + dummy->pcm_hw.periods_min = m->periods_min; + if (m->periods_max) + dummy->pcm_hw.periods_max = m->periods_max; + if (m->rates) + dummy->pcm_hw.rates = m->rates; + if (m->rate_min) + dummy->pcm_hw.rate_min = m->rate_min; + if (m->rate_max) + dummy->pcm_hw.rate_max = m->rate_max; + if (m->channels_min) + dummy->pcm_hw.channels_min = m->channels_min; + if (m->channels_max) + dummy->pcm_hw.channels_max = m->channels_max; + } + err = snd_card_dummy_new_mixer(dummy); if (err < 0) goto __nodev; -- cgit v0.10.2 From 0f69d9782c6e6a7b0e60113a850845bc642c3f4e Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 3 Feb 2010 17:37:23 +0100 Subject: ASoC: fix compilation breakage in sound/soc/sh/fsi.c ctrl_outl() has become void at some point, which breaks compilation of fsi.c. Make writing functions void, as their output is anyway not evaluated, and use __raw_writel and __raw_readl instead of deprecated ctrl_outl and ctrl_inl respectively. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ebf3588..3c36d24 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -120,35 +120,35 @@ struct fsi_master { ************************************************************************/ -static int __fsi_reg_write(u32 reg, u32 data) +static void __fsi_reg_write(u32 reg, u32 data) { /* valid data area is 24bit */ data &= 0x00ffffff; - return ctrl_outl(data, reg); + __raw_writel(data, reg); } static u32 __fsi_reg_read(u32 reg) { - return ctrl_inl(reg); + return __raw_readl(reg); } -static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) { u32 val = __fsi_reg_read(reg); val &= ~mask; val |= data & mask; - return __fsi_reg_write(reg, val); + __fsi_reg_write(reg, val); } -static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +static void fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) { if (reg > REG_END) - return -1; + return; - return __fsi_reg_write((u32)(fsi->base + reg), data); + __fsi_reg_write((u32)(fsi->base + reg), data); } static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) @@ -159,28 +159,25 @@ static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) return __fsi_reg_read((u32)(fsi->base + reg)); } -static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +static void fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) { if (reg > REG_END) - return -1; + return; - return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); + __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); } -static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) +static void fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { - int ret; unsigned long flags; if ((reg < MREG_START) || (reg > MREG_END)) - return -1; + return; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_write((u32)(master->base + reg), data); + __fsi_reg_write((u32)(master->base + reg), data); spin_unlock_irqrestore(&master->lock, flags); - - return ret; } static u32 fsi_master_read(struct fsi_master *master, u32 reg) @@ -199,21 +196,18 @@ static u32 fsi_master_read(struct fsi_master *master, u32 reg) return ret; } -static int fsi_master_mask_set(struct fsi_master *master, +static void fsi_master_mask_set(struct fsi_master *master, u32 reg, u32 mask, u32 data) { - int ret; unsigned long flags; if ((reg < MREG_START) || (reg > MREG_END)) - return -1; + return; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + __fsi_reg_mask_set((u32)(master->base + reg), mask, data); spin_unlock_irqrestore(&master->lock, flags); - - return ret; } /************************************************************************ -- cgit v0.10.2 From 8c961bcca1d10be4f2c06375eb561679167653a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Feb 2010 18:46:10 +0000 Subject: ASoC: Allow CODECs to ask soc-cache to suppress physical writes Currently the soc-cache code will always write to the device, meaning that we need the device to be powered and active at pretty much all times the system is active. Allowing cache only writes lays some groundwork for future enhancements to allow devices to be put into a full off state when the audio subsystem is idle. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/soc.h b/include/sound/soc.h index 4bbeb9f..4e8f14b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -423,6 +423,7 @@ struct snd_soc_codec { short reg_cache_step; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + unsigned int cache_only:1; /* Suppress writes to hardware */ /* dapm */ u32 pop_time; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 097e335..84b6916 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -38,6 +38,10 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -100,6 +104,10 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -153,6 +161,9 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + if (codec->cache_only) + return 0; + if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -181,6 +192,9 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, if (!snd_soc_codec_volatile_register(codec, reg)) reg_cache[reg] = value; + if (codec->cache_only) + return 0; + if (codec->hw_write(codec->control_data, data, 3) == 3) return 0; else @@ -193,10 +207,14 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; if (reg >= codec->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) + snd_soc_codec_volatile_register(codec, reg)) { + if (codec->cache_only) + return -EINVAL; + return codec->hw_read(codec, reg); - else + } else { return cache[reg]; + } } #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) @@ -294,6 +312,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 3); if (ret == 3) return 0; -- cgit v0.10.2 From a9694faa287888b4fb10849649b6c94d0a1c9940 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Thu, 4 Feb 2010 08:58:23 +0100 Subject: ALSA: hda - Adding support for another IDT 92HD83XXX codec Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9694675..693dd14 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5341,6 +5341,7 @@ again: spec->num_pwrs = 0; break; case 0x111d7604: + case 0x111d76d4: case 0x111d7605: case 0x111d76d5: if (spec->board_config == STAC_92HD83XXX_PWR_REF) @@ -6263,6 +6264,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d4, .name = "92HD83C1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx}, -- cgit v0.10.2 From 04b5efe5fa7f71c37b938053666fac317b67c636 Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Thu, 4 Feb 2010 10:28:02 +0100 Subject: ALSA: hda - Fix docking output for IDT 92HD8xx codecs This patch fixes docking output support for IDT 92HD81/83/88 family codecs. Typically one of ports 0xE or 0xF is used for docking output, while only port 0xF is common on all the three codec families. We don't want the pin to select the analog mixer here. Signed-off-by: Charles Chin Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 693dd14..834c598 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5291,7 +5291,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; int num_dacs; - hda_nid_t nid; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -5387,24 +5386,21 @@ again: return err; } - switch (spec->board_config) { - case STAC_DELL_S14: - nid = 0xf; - break; - default: - nid = 0xe; - break; - } - - num_dacs = snd_hda_get_connections(codec, nid, + /* docking output support */ + num_dacs = snd_hda_get_connections(codec, 0xF, conn, STAC92HD83_DAC_COUNT + 1) - 1; - if (num_dacs < 0) - num_dacs = STAC92HD83_DAC_COUNT; - - /* set port X to select the last DAC - */ - snd_hda_codec_write_cache(codec, nid, 0, + /* skip non-DAC connections */ + while (num_dacs >= 0 && + (get_wcaps_type(get_wcaps(codec, conn[num_dacs])) + != AC_WID_AUD_OUT)) + num_dacs--; + /* set port E and F to select the last DAC */ + if (num_dacs >= 0) { + snd_hda_codec_write_cache(codec, 0xE, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + snd_hda_codec_write_cache(codec, 0xF, 0, AC_VERB_SET_CONNECT_SEL, num_dacs); + } codec->proc_widget_hook = stac92hd_proc_hook; -- cgit v0.10.2 From a3032b47c46920ed3f2fd58e64f484e3dab49f23 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Feb 2010 18:48:03 +0000 Subject: ASoC: Add a cache_sync bit to the CODEC structure Add a bit to the CODEC structure indicating if a cache sync is required. By default this will be set if a cache only write is done to a soc-cache register cache. This allows us to avoid syncing the cache back after using cache only writes if there were no changes. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/soc.h b/include/sound/soc.h index 4e8f14b..e6a6d10 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -424,6 +424,7 @@ struct snd_soc_codec { unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ unsigned int cache_only:1; /* Suppress writes to hardware */ + unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ /* dapm */ u32 pop_time; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 84b6916..5869dc3 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -39,8 +39,10 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) @@ -105,8 +107,10 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) @@ -161,8 +165,10 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; @@ -192,8 +198,10 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, if (!snd_soc_codec_volatile_register(codec, reg)) reg_cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } if (codec->hw_write(codec->control_data, data, 3) == 3) return 0; @@ -313,8 +321,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; - if (codec->cache_only) + if (codec->cache_only) { + codec->cache_sync = 1; return 0; + } ret = codec->hw_write(codec->control_data, data, 3); if (ret == 3) -- cgit v0.10.2 From 3bf6e4217e3c69438f6dc41a009664107eb27ab1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Feb 2010 19:05:09 +0000 Subject: ASoC: Convert WM8993 to use shared cache I/O code Saves a little bit of code duplication. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 61239e0..3c9336c 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -231,34 +231,6 @@ struct wm8993_priv { int fll_src; }; -static unsigned int wm8993_read_hw(struct snd_soc_codec *codec, u8 reg) -{ - struct i2c_msg xfer[2]; - u16 data; - int ret; - struct i2c_client *i2c = codec->control_data; - - /* Write register */ - xfer[0].addr = i2c->addr; - xfer[0].flags = 0; - xfer[0].len = 1; - xfer[0].buf = ® - - /* Read data */ - xfer[1].addr = i2c->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 2; - xfer[1].buf = (u8 *)&data; - - ret = i2c_transfer(i2c->adapter, xfer, 2); - if (ret != 2) { - dev_err(codec->dev, "Failed to read 0x%x: %d\n", reg, ret); - return 0; - } - - return (data >> 8) | ((data & 0xff) << 8); -} - static int wm8993_volatile(unsigned int reg) { switch (reg) { @@ -273,48 +245,6 @@ static int wm8993_volatile(unsigned int reg) } } -static unsigned int wm8993_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *reg_cache = codec->reg_cache; - - BUG_ON(reg > WM8993_MAX_REGISTER); - - if (wm8993_volatile(reg)) - return wm8993_read_hw(codec, reg); - else - return reg_cache[reg]; -} - -static int wm8993_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 *reg_cache = codec->reg_cache; - u8 data[3]; - int ret; - - BUG_ON(reg > WM8993_MAX_REGISTER); - - /* data is - * D15..D9 WM8993 register offset - * D8...D0 register data - */ - data[0] = reg; - data[1] = value >> 8; - data[2] = value & 0x00ff; - - if (!wm8993_volatile(reg)) - reg_cache[reg] = value; - - ret = codec->hw_write(codec->control_data, data, 3); - - if (ret == 3) - return 0; - if (ret < 0) - return ret; - return -EIO; -} - struct _fll_div { u16 fll_fratio; u16 fll_outdiv; @@ -443,9 +373,9 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, wm8993->fll_fref = 0; wm8993->fll_fout = 0; - reg1 = wm8993_read(codec, WM8993_FLL_CONTROL_1); + reg1 = snd_soc_read(codec, WM8993_FLL_CONTROL_1); reg1 &= ~WM8993_FLL_ENA; - wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1); return 0; } @@ -454,7 +384,7 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, if (ret != 0) return ret; - reg5 = wm8993_read(codec, WM8993_FLL_CONTROL_5); + reg5 = snd_soc_read(codec, WM8993_FLL_CONTROL_5); reg5 &= ~WM8993_FLL_CLK_SRC_MASK; switch (fll_id) { @@ -476,33 +406,33 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, /* Any FLL configuration change requires that the FLL be * disabled first. */ - reg1 = wm8993_read(codec, WM8993_FLL_CONTROL_1); + reg1 = snd_soc_read(codec, WM8993_FLL_CONTROL_1); reg1 &= ~WM8993_FLL_ENA; - wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1); /* Apply the configuration */ if (fll_div.k) reg1 |= WM8993_FLL_FRAC_MASK; else reg1 &= ~WM8993_FLL_FRAC_MASK; - wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1); - wm8993_write(codec, WM8993_FLL_CONTROL_2, - (fll_div.fll_outdiv << WM8993_FLL_OUTDIV_SHIFT) | - (fll_div.fll_fratio << WM8993_FLL_FRATIO_SHIFT)); - wm8993_write(codec, WM8993_FLL_CONTROL_3, fll_div.k); + snd_soc_write(codec, WM8993_FLL_CONTROL_2, + (fll_div.fll_outdiv << WM8993_FLL_OUTDIV_SHIFT) | + (fll_div.fll_fratio << WM8993_FLL_FRATIO_SHIFT)); + snd_soc_write(codec, WM8993_FLL_CONTROL_3, fll_div.k); - reg4 = wm8993_read(codec, WM8993_FLL_CONTROL_4); + reg4 = snd_soc_read(codec, WM8993_FLL_CONTROL_4); reg4 &= ~WM8993_FLL_N_MASK; reg4 |= fll_div.n << WM8993_FLL_N_SHIFT; - wm8993_write(codec, WM8993_FLL_CONTROL_4, reg4); + snd_soc_write(codec, WM8993_FLL_CONTROL_4, reg4); reg5 &= ~WM8993_FLL_CLK_REF_DIV_MASK; reg5 |= fll_div.fll_clk_ref_div << WM8993_FLL_CLK_REF_DIV_SHIFT; - wm8993_write(codec, WM8993_FLL_CONTROL_5, reg5); + snd_soc_write(codec, WM8993_FLL_CONTROL_5, reg5); /* Enable the FLL */ - wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); + snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); @@ -523,7 +453,7 @@ static int configure_clock(struct snd_soc_codec *codec) case WM8993_SYSCLK_MCLK: dev_dbg(codec->dev, "Using %dHz MCLK\n", wm8993->mclk_rate); - reg = wm8993_read(codec, WM8993_CLOCKING_2); + reg = snd_soc_read(codec, WM8993_CLOCKING_2); reg &= ~(WM8993_MCLK_DIV | WM8993_SYSCLK_SRC); if (wm8993->mclk_rate > 13500000) { reg |= WM8993_MCLK_DIV; @@ -532,14 +462,14 @@ static int configure_clock(struct snd_soc_codec *codec) reg &= ~WM8993_MCLK_DIV; wm8993->sysclk_rate = wm8993->mclk_rate; } - wm8993_write(codec, WM8993_CLOCKING_2, reg); + snd_soc_write(codec, WM8993_CLOCKING_2, reg); break; case WM8993_SYSCLK_FLL: dev_dbg(codec->dev, "Using %dHz FLL clock\n", wm8993->fll_fout); - reg = wm8993_read(codec, WM8993_CLOCKING_2); + reg = snd_soc_read(codec, WM8993_CLOCKING_2); reg |= WM8993_SYSCLK_SRC; if (wm8993->fll_fout > 13500000) { reg |= WM8993_MCLK_DIV; @@ -548,7 +478,7 @@ static int configure_clock(struct snd_soc_codec *codec) reg &= ~WM8993_MCLK_DIV; wm8993->sysclk_rate = wm8993->fll_fout; } - wm8993_write(codec, WM8993_CLOCKING_2, reg); + snd_soc_write(codec, WM8993_CLOCKING_2, reg); break; default: @@ -1083,8 +1013,8 @@ static int wm8993_set_dai_fmt(struct snd_soc_dai *dai, { struct snd_soc_codec *codec = dai->codec; struct wm8993_priv *wm8993 = codec->private_data; - unsigned int aif1 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_1); - unsigned int aif4 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_4); + unsigned int aif1 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_1); + unsigned int aif4 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_4); aif1 &= ~(WM8993_BCLK_DIR | WM8993_AIF_BCLK_INV | WM8993_AIF_LRCLK_INV | WM8993_AIF_FMT_MASK); @@ -1167,8 +1097,8 @@ static int wm8993_set_dai_fmt(struct snd_soc_dai *dai, return -EINVAL; } - wm8993_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); - wm8993_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); + snd_soc_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); + snd_soc_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); return 0; } @@ -1182,16 +1112,16 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream, int ret, i, best, best_val, cur_val; unsigned int clocking1, clocking3, aif1, aif4; - clocking1 = wm8993_read(codec, WM8993_CLOCKING_1); + clocking1 = snd_soc_read(codec, WM8993_CLOCKING_1); clocking1 &= ~WM8993_BCLK_DIV_MASK; - clocking3 = wm8993_read(codec, WM8993_CLOCKING_3); + clocking3 = snd_soc_read(codec, WM8993_CLOCKING_3); clocking3 &= ~(WM8993_CLK_SYS_RATE_MASK | WM8993_SAMPLE_RATE_MASK); - aif1 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_1); + aif1 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_1); aif1 &= ~WM8993_AIF_WL_MASK; - aif4 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_4); + aif4 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_4); aif4 &= ~WM8993_LRCLK_RATE_MASK; /* What BCLK do we need? */ @@ -1284,14 +1214,14 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm8993->bclk / wm8993->fs); aif4 |= wm8993->bclk / wm8993->fs; - wm8993_write(codec, WM8993_CLOCKING_1, clocking1); - wm8993_write(codec, WM8993_CLOCKING_3, clocking3); - wm8993_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); - wm8993_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); + snd_soc_write(codec, WM8993_CLOCKING_1, clocking1); + snd_soc_write(codec, WM8993_CLOCKING_3, clocking3); + snd_soc_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); + snd_soc_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); /* ReTune Mobile? */ if (wm8993->pdata.num_retune_configs) { - u16 eq1 = wm8993_read(codec, WM8993_EQ1); + u16 eq1 = snd_soc_read(codec, WM8993_EQ1); struct wm8993_retune_mobile_setting *s; best = 0; @@ -1314,7 +1244,7 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, WM8993_EQ1, WM8993_EQ_ENA, 0); for (i = 1; i < ARRAY_SIZE(s->config); i++) - wm8993_write(codec, WM8993_EQ1 + i, s->config[i]); + snd_soc_write(codec, WM8993_EQ1 + i, s->config[i]); snd_soc_update_bits(codec, WM8993_EQ1, WM8993_EQ_ENA, eq1); } @@ -1327,14 +1257,14 @@ static int wm8993_digital_mute(struct snd_soc_dai *codec_dai, int mute) struct snd_soc_codec *codec = codec_dai->codec; unsigned int reg; - reg = wm8993_read(codec, WM8993_DAC_CTRL); + reg = snd_soc_read(codec, WM8993_DAC_CTRL); if (mute) reg |= WM8993_DAC_MUTE; else reg &= ~WM8993_DAC_MUTE; - wm8993_write(codec, WM8993_DAC_CTRL, reg); + snd_soc_write(codec, WM8993_DAC_CTRL, reg); return 0; } @@ -1586,9 +1516,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, INIT_LIST_HEAD(&codec->dapm_paths); codec->name = "WM8993"; - codec->read = wm8993_read; - codec->write = wm8993_write; - codec->hw_write = (hw_write_t)i2c_master_send; + codec->volatile_register = wm8993_volatile; codec->reg_cache = wm8993->reg_cache; codec->reg_cache_size = ARRAY_SIZE(wm8993->reg_cache); codec->bias_level = SND_SOC_BIAS_OFF; @@ -1603,20 +1531,26 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, memcpy(wm8993->reg_cache, wm8993_reg_defaults, sizeof(wm8993->reg_cache)); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + i2c_set_clientdata(i2c, wm8993); codec->control_data = i2c; wm8993_codec = codec; codec->dev = &i2c->dev; - val = wm8993_read_hw(codec, WM8993_SOFTWARE_RESET); + val = snd_soc_read(codec, WM8993_SOFTWARE_RESET); if (val != wm8993_reg_defaults[WM8993_SOFTWARE_RESET]) { dev_err(codec->dev, "Invalid ID register value %x\n", val); ret = -EINVAL; goto err; } - ret = wm8993_write(codec, WM8993_SOFTWARE_RESET, 0xffff); + ret = snd_soc_write(codec, WM8993_SOFTWARE_RESET, 0xffff); if (ret != 0) goto err; -- cgit v0.10.2 From b37e399bfc7fcb5b523e3e2e74686c8cc95c0cba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Feb 2010 11:51:42 +0000 Subject: ASoC: Initial WM8993 regulator API hookup At the minute the regulators are simply enabled for the entire lifetime of the device. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 3c9336c..e97b3f4 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -29,6 +30,16 @@ #include "wm8993.h" #include "wm_hubs.h" +#define WM8993_NUM_SUPPLIES 6 +static const char *wm8993_supply_names[WM8993_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD", + "AVDD1", + "AVDD2", + "CPVDD", + "SPKVDD", +}; + static u16 wm8993_reg_defaults[WM8993_REGISTER_COUNT] = { 0x8993, /* R0 - Software Reset */ 0x0000, /* R1 - Power Management (1) */ @@ -215,6 +226,7 @@ static struct { struct wm8993_priv { struct wm_hubs_data hubs_data; u16 reg_cache[WM8993_REGISTER_COUNT]; + struct regulator_bulk_data supplies[WM8993_NUM_SUPPLIES]; struct wm8993_platform_data pdata; struct snd_soc_codec codec; int master; @@ -1496,6 +1508,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, struct snd_soc_codec *codec; unsigned int val; int ret; + int i; if (wm8993_codec) { dev_err(&i2c->dev, "A WM8993 is already registered\n"); @@ -1543,16 +1556,33 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, codec->dev = &i2c->dev; + for (i = 0; i < ARRAY_SIZE(wm8993->supplies); i++) + wm8993->supplies[i].supply = wm8993_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + val = snd_soc_read(codec, WM8993_SOFTWARE_RESET); if (val != wm8993_reg_defaults[WM8993_SOFTWARE_RESET]) { dev_err(codec->dev, "Invalid ID register value %x\n", val); ret = -EINVAL; - goto err; + goto err_enable; } ret = snd_soc_write(codec, WM8993_SOFTWARE_RESET, 0xffff); if (ret != 0) - goto err; + goto err_enable; /* By default we're using the output mixers */ wm8993->class_w_users = 2; @@ -1582,7 +1612,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret != 0) - goto err; + goto err_enable; wm8993_dai.dev = codec->dev; @@ -1596,6 +1626,10 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, err_bias: wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); err: wm8993_codec = NULL; kfree(wm8993); @@ -1610,6 +1644,7 @@ static int wm8993_i2c_remove(struct i2c_client *client) snd_soc_unregister_dai(&wm8993_dai); wm8993_set_bias_level(&wm8993->codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); kfree(wm8993); return 0; -- cgit v0.10.2 From cf56f62746c3e2f70bfad3d6fd051427a0022368 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Feb 2010 17:55:55 +0000 Subject: ASoC: Disable WM8993 regulators when turning bias off While the regulators are disabled we cache all register writes. Currently we assume that the regulator disable actually takes effect, after the merge with the regulator tree in 2.6.34 the regulator API will be able to notify us if the power is actually removed (due to constraints or regulator sharing it may not be). Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index e97b3f4..bf022f6 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -923,10 +923,33 @@ static const struct snd_soc_dapm_route routes[] = { { "Right Headphone Mux", "DAC", "DACR" }, }; +static void wm8993_cache_restore(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i; + + if (!codec->cache_sync) + return; + + /* Reenable hardware writes */ + codec->cache_only = 0; + + /* Restore the register settings */ + for (i = 1; i < WM8993_MAX_REGISTER; i++) { + if (cache[i] == wm8993_reg_defaults[i]) + continue; + snd_soc_write(codec, i, cache[i]); + } + + /* We're in sync again */ + codec->cache_sync = 0; +} + static int wm8993_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8993_priv *wm8993 = codec->private_data; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -940,6 +963,13 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); + if (ret != 0) + return ret; + + wm8993_cache_restore(codec); + /* Tune DC servo configuration */ snd_soc_write(codec, 0x44, 3); snd_soc_write(codec, 0x56, 3); @@ -992,6 +1022,18 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, WM8993_VMID_SEL_MASK | WM8993_BIAS_ENA, 0); + +#ifdef CONFIG_REGULATOR + /* Post 2.6.34 we will be able to get a callback when + * the regulators are disabled which we can use but + * for now just assume that the power will be cut if + * the regulator API is in use. + */ + codec->cache_sync = 1; +#endif + + regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); break; } @@ -1460,15 +1502,7 @@ static int wm8993_resume(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; struct wm8993_priv *wm8993 = codec->private_data; - u16 *cache = wm8993->reg_cache; - int i, ret; - - /* Restore the register settings */ - for (i = 1; i < WM8993_MAX_REGISTER; i++) { - if (cache[i] == wm8993_reg_defaults[i]) - continue; - snd_soc_write(codec, i, cache[i]); - } + int ret; wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1584,6 +1618,8 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_enable; + codec->cache_only = 1; + /* By default we're using the output mixers */ wm8993->class_w_users = 2; -- cgit v0.10.2 From c133421800d9d1dfec0c98de6c9da0a7a99e0573 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Jan 2010 22:37:11 +0000 Subject: ASoC: Add support for BIAS_OFF when idle to WM8904 As well as disabling the biases of the CODEC the drop into BIAS_OFF will also disable all the regulators powering the CODEC, allowing even greater power savings on appropriately configured systems. Since the regulator API does not currently provide notification when regulators are disabled we assume that this always happens when we stop using the regulators. Once 2.6.34 is merged this code can be optimised to only sync the cache when power was actually removed. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 992a7f2..dc782c4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2033,11 +2033,37 @@ static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static void wm8904_sync_cache(struct snd_soc_codec *codec) +{ + struct wm8904_priv *wm8904 = codec->private_data; + int i; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + + /* Sync back cached values if they're different from the + * hardware default. + */ + for (i = 1; i < ARRAY_SIZE(wm8904->reg_cache); i++) { + if (!wm8904_access[i].writable) + continue; + + if (wm8904->reg_cache[i] == wm8904_reg[i]) + continue; + + snd_soc_write(codec, i, wm8904->reg_cache[i]); + } + + codec->cache_sync = 0; +} + static int wm8904_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8904_priv *wm8904 = codec->private_data; - int ret, i; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -2065,18 +2091,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, return ret; } - /* Sync back cached values if they're - * different from the hardware default. - */ - for (i = 1; i < ARRAY_SIZE(wm8904->reg_cache); i++) { - if (!wm8904_access[i].writable) - continue; - - if (wm8904->reg_cache[i] == wm8904_reg[i]) - continue; - - snd_soc_write(codec, i, wm8904->reg_cache[i]); - } + wm8904_sync_cache(codec); /* Enable bias */ snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, @@ -2112,6 +2127,15 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); +#ifdef CONFIG_REGULATOR + /* Post 2.6.34 we will be able to get a callback when + * the regulators are disabled which we can use but + * for now just assume that the power will be cut if + * the regulator API is in use. + */ + codec->cache_sync = 1; +#endif + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); break; @@ -2365,6 +2389,8 @@ static int wm8904_register(struct wm8904_priv *wm8904, codec->reg_cache_size = WM8904_MAX_REGISTER; codec->reg_cache = &wm8904->reg_cache; codec->volatile_register = wm8904_volatile_register; + codec->cache_sync = 1; + codec->idle_bias_off = 1; memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg)); -- cgit v0.10.2 From e4bc669610d75106a00b0f96f2410ac5898ef1ca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Feb 2010 19:51:33 +0000 Subject: ASoC: Optimise WM8904 output stage power control Handle the output PGAs as part of the output powerup since they can never be powered separately and reorder things so that we remove the output shorts after both line and headphone outputs have been brought up, minimising the opportunity for any issues. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index dc782c4..80dd8df 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -979,6 +979,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, int dcs_l, dcs_r; int dcs_l_reg, dcs_r_reg; int timeout; + int pwr_reg; /* This code is shared between HP and LINEOUT; we do all our * power management in stereo pairs to avoid latency issues so @@ -988,6 +989,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, switch (reg) { case WM8904_ANALOGUE_HP_0: + pwr_reg = WM8904_POWER_MANAGEMENT_2; dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1; dcs_r_reg = WM8904_DC_SERVO_8; dcs_l_reg = WM8904_DC_SERVO_9; @@ -995,6 +997,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_r = 1; break; case WM8904_ANALOGUE_LINEOUT_0: + pwr_reg = WM8904_POWER_MANAGEMENT_3; dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3; dcs_r_reg = WM8904_DC_SERVO_6; dcs_l_reg = WM8904_DC_SERVO_7; @@ -1007,12 +1010,18 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, } switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: + /* Power on the PGAs */ + snd_soc_update_bits(codec, pwr_reg, + WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA, + WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA); + /* Power on the amplifier */ snd_soc_update_bits(codec, reg, WM8904_HPL_ENA | WM8904_HPR_ENA, WM8904_HPL_ENA | WM8904_HPR_ENA); + /* Enable the first stage */ snd_soc_update_bits(codec, reg, WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY, @@ -1064,7 +1073,9 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, reg, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP); + break; + case SND_SOC_DAPM_POST_PMU: /* Unshort the output itself */ snd_soc_update_bits(codec, reg, WM8904_HPL_RMV_SHORT | @@ -1079,7 +1090,9 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, reg, WM8904_HPL_RMV_SHORT | WM8904_HPR_RMV_SHORT, 0); + break; + case SND_SOC_DAPM_POST_PMD: /* Cache the DC servo configuration; this will be * invalidated if we change the configuration. */ wm8904->dcs_state[dcs_l] = snd_soc_read(codec, dcs_l_reg); @@ -1094,6 +1107,11 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY | WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, 0); + + /* PGAs too */ + snd_soc_update_bits(codec, pwr_reg, + WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA, + 0); break; } @@ -1212,18 +1230,20 @@ SND_SOC_DAPM_DAC("DACR", NULL, WM8904_POWER_MANAGEMENT_6, 2, 0), SND_SOC_DAPM_SUPPLY("Charge pump", WM8904_CHARGE_PUMP_0, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("HPL PGA", WM8904_POWER_MANAGEMENT_2, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA("HPR PGA", WM8904_POWER_MANAGEMENT_2, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("HPL PGA", SND_SOC_NOPM, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("HPR PGA", SND_SOC_NOPM, 0, 0, NULL, 0), -SND_SOC_DAPM_PGA("LINEL PGA", WM8904_POWER_MANAGEMENT_3, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA("LINER PGA", WM8904_POWER_MANAGEMENT_3, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("LINEL PGA", SND_SOC_NOPM, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("LINER PGA", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM, WM8904_ANALOGUE_HP_0, 0, NULL, 0, out_pga_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA_E("Line Output", SND_SOC_NOPM, WM8904_ANALOGUE_LINEOUT_0, 0, NULL, 0, out_pga_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), -- cgit v0.10.2 From 8c1264740e7c9688c5d11b96d26e4393618ef60e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Feb 2010 19:33:49 +0000 Subject: ASoC: Add WM8912 DAC support The WM8912 is a DAC only device register compatible with the WM8904 CODEC with ADC portions omitted. Support it within the WM8904 driver based on the configured I2C device name. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 80dd8df..593e47d 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -33,6 +33,11 @@ static struct snd_soc_codec *wm8904_codec; struct snd_soc_codec_device soc_codec_dev_wm8904; +enum wm8904_type { + WM8904, + WM8912, +}; + #define WM8904_NUM_DCS_CHANNELS 4 #define WM8904_NUM_SUPPLIES 5 @@ -49,6 +54,8 @@ struct wm8904_priv { struct snd_soc_codec codec; u16 reg_cache[WM8904_MAX_REGISTER + 1]; + enum wm8904_type devtype; + struct regulator_bulk_data supplies[WM8904_NUM_SUPPLIES]; struct wm8904_pdata *pdata; @@ -1411,30 +1418,62 @@ static const struct snd_soc_dapm_route wm8904_intercon[] = { { "LINER PGA", NULL, "LINER Mux" }, }; +static const struct snd_soc_dapm_route wm8912_intercon[] = { + { "HPL PGA", NULL, "DACL" }, + { "HPR PGA", NULL, "DACR" }, + + { "LINEL PGA", NULL, "DACL" }, + { "LINER PGA", NULL, "DACR" }, +}; + static int wm8904_add_widgets(struct snd_soc_codec *codec) { - snd_soc_add_controls(codec, wm8904_adc_snd_controls, - ARRAY_SIZE(wm8904_adc_snd_controls)); - snd_soc_add_controls(codec, wm8904_dac_snd_controls, - ARRAY_SIZE(wm8904_dac_snd_controls)); - snd_soc_add_controls(codec, wm8904_snd_controls, - ARRAY_SIZE(wm8904_snd_controls)); + struct wm8904_priv *wm8904 = codec->private_data; snd_soc_dapm_new_controls(codec, wm8904_core_dapm_widgets, ARRAY_SIZE(wm8904_core_dapm_widgets)); - snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets, - ARRAY_SIZE(wm8904_adc_dapm_widgets)); - snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, - ARRAY_SIZE(wm8904_dac_dapm_widgets)); - snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets, - ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(codec, core_intercon, ARRAY_SIZE(core_intercon)); - snd_soc_dapm_add_routes(codec, adc_intercon, ARRAY_SIZE(adc_intercon)); - snd_soc_dapm_add_routes(codec, dac_intercon, ARRAY_SIZE(dac_intercon)); - snd_soc_dapm_add_routes(codec, wm8904_intercon, - ARRAY_SIZE(wm8904_intercon)); + + switch (wm8904->devtype) { + case WM8904: + snd_soc_add_controls(codec, wm8904_adc_snd_controls, + ARRAY_SIZE(wm8904_adc_snd_controls)); + snd_soc_add_controls(codec, wm8904_dac_snd_controls, + ARRAY_SIZE(wm8904_dac_snd_controls)); + snd_soc_add_controls(codec, wm8904_snd_controls, + ARRAY_SIZE(wm8904_snd_controls)); + + snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets, + ARRAY_SIZE(wm8904_adc_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, + ARRAY_SIZE(wm8904_dac_dapm_widgets)); + snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets, + ARRAY_SIZE(wm8904_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, core_intercon, + ARRAY_SIZE(core_intercon)); + snd_soc_dapm_add_routes(codec, adc_intercon, + ARRAY_SIZE(adc_intercon)); + snd_soc_dapm_add_routes(codec, dac_intercon, + ARRAY_SIZE(dac_intercon)); + snd_soc_dapm_add_routes(codec, wm8904_intercon, + ARRAY_SIZE(wm8904_intercon)); + break; + + case WM8912: + snd_soc_add_controls(codec, wm8904_dac_snd_controls, + ARRAY_SIZE(wm8904_dac_snd_controls)); + + snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, + ARRAY_SIZE(wm8904_dac_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, dac_intercon, + ARRAY_SIZE(dac_intercon)); + snd_soc_dapm_add_routes(codec, wm8912_intercon, + ARRAY_SIZE(wm8912_intercon)); + break; + } snd_soc_dapm_new_widgets(codec); return 0; @@ -2412,6 +2451,18 @@ static int wm8904_register(struct wm8904_priv *wm8904, codec->cache_sync = 1; codec->idle_bias_off = 1; + switch (wm8904->devtype) { + case WM8904: + break; + case WM8912: + memset(&wm8904_dai.capture, 0, sizeof(wm8904_dai.capture)); + break; + default: + dev_err(codec->dev, "Unknown device type %d\n", + wm8904->devtype); + return -EINVAL; + } + memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg)); ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); @@ -2542,6 +2593,8 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, codec = &wm8904->codec; codec->hw_write = (hw_write_t)i2c_master_send; + wm8904->devtype = id->driver_data; + i2c_set_clientdata(i2c, wm8904); codec->control_data = i2c; wm8904->pdata = i2c->dev.platform_data; @@ -2559,7 +2612,8 @@ static __devexit int wm8904_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8904_i2c_id[] = { - { "wm8904", 0 }, + { "wm8904", WM8904 }, + { "wm8912", WM8912 }, { } }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); -- cgit v0.10.2 From cb67286d6629ecb5bfc44071d664cf1cbd01a350 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 4 Feb 2010 09:10:10 +0200 Subject: ASoC: TWL4030: Module unloading fix The module unloading path had several problems: - it freed up the private structure twice - it freed up the codec structure, which was allocated as part of the private structure - it did not freed up the reg_cache - it did not unregistered the dais and the codec Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e0106a5..b32aeb3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2152,8 +2152,6 @@ static int twl4030_soc_remove(struct platform_device *pdev) twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - kfree(codec->private_data); - kfree(codec); return 0; } @@ -2237,6 +2235,9 @@ static int __devexit twl4030_codec_remove(struct platform_device *pdev) { struct twl4030_priv *twl4030 = platform_get_drvdata(pdev); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + snd_soc_unregister_codec(&twl4030->codec); + kfree(twl4030->codec.reg_cache); kfree(twl4030); twl4030_codec = NULL; -- cgit v0.10.2 From 88102f3f841b680412714d0b0b7da33c2a00c1f9 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 4 Feb 2010 14:12:58 +0100 Subject: ALSA: hda - Remove superfluous init verb entries for ALC88[235] The default values are no need to be set in init_verbs. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b2f543d..40ebf27 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7332,29 +7332,18 @@ static struct snd_kcontrol_new alc882_chmode_mixer[] = { static struct hda_verb alc882_base_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* mute analog input loopbacks */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Front Pin: output 0 (0x0c) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -7391,14 +7380,8 @@ static struct hda_verb alc882_base_init_verbs[] = { /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* Input mixer3 */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* ADC2: mute amp left and right */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -7442,26 +7425,17 @@ static struct hda_verb alc_hp15_unsol_verbs[] = { static struct hda_verb alc885_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* mute analog input loopbacks */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Front HP Pin: output 0 (0x0c) */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -7495,17 +7469,11 @@ static struct hda_verb alc885_init_verbs[] = { /* Mixer elements: 0x18, , 0x1a, 0x1b */ /* Input mixer1 */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* ADC2: mute amp left and right */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* ADC3: mute amp left and right */ @@ -7991,18 +7959,6 @@ static struct hda_verb alc883_auto_init_verbs[] = { {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* * Set up output mixers (0x0c - 0x0f) */ @@ -8027,16 +7983,9 @@ static struct hda_verb alc883_auto_init_verbs[] = { /* FIXME: use matrix-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { } }; -- cgit v0.10.2 From 84898e87cc0fff976202d5b91656f2db949fc2dd Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 4 Feb 2010 14:16:14 +0100 Subject: ALSA: hda - Add ALC269VB support - Add new models ALC269VB_AMIC ALC269VB_DMIC - Add alc269vb_laptop_dmic_setup The record source index Dmic is 0x6 for ALC269VB. - Change eeepc words for ALC269 - Modify init_verb tables of patch_alc269 patch_alc662 patch_alc882 - Modify common patch for ALC270 ALC269VB ALC275 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 40ebf27..826ecdb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -131,8 +131,10 @@ enum { enum { ALC269_BASIC, ALC269_QUANTA_FL1, - ALC269_ASUS_AMIC, - ALC269_ASUS_DMIC, + ALC269_AMIC, + ALC269_DMIC, + ALC269VB_AMIC, + ALC269VB_DMIC, ALC269_FUJITSU, ALC269_LIFEBOOK, ALC269_AUTO, @@ -13182,6 +13184,15 @@ static hda_nid_t alc269_capsrc_nids[1] = { 0x23, }; +static hda_nid_t alc269vb_adc_nids[1] = { + /* ADC1 */ + 0x09, +}; + +static hda_nid_t alc269vb_capsrc_nids[1] = { + 0x22, +}; + /* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24), * not a mux! */ @@ -13250,7 +13261,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { } }; -static struct snd_kcontrol_new alc269_eeepc_mixer[] = { +static struct snd_kcontrol_new alc269_laptop_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -13258,16 +13269,47 @@ static struct snd_kcontrol_new alc269_eeepc_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc269vb_laptop_mixer[] = { + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + { } /* end */ +}; + /* capture mixer elements */ -static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { +static struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("IntMic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("IntMic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), { } /* end */ }; /* FSC amilo */ -#define alc269_fujitsu_mixer alc269_eeepc_mixer +#define alc269_fujitsu_mixer alc269_laptop_mixer static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -13410,7 +13452,7 @@ static void alc269_lifebook_init_hook(struct hda_codec *codec) alc269_lifebook_mic_autoswitch(codec); } -static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { +static struct hda_verb alc269_laptop_dmic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, @@ -13421,7 +13463,7 @@ static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { {} }; -static struct hda_verb alc269_eeepc_amic_init_verbs[] = { +static struct hda_verb alc269_laptop_amic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, @@ -13431,6 +13473,28 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = { {} }; +static struct hda_verb alc269vb_laptop_dmic_init_verbs[] = { + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x06}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc269vb_laptop_amic_init_verbs[] = { + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + /* toggle speaker-output according to the hp-jack state */ static void alc269_speaker_automute(struct hda_codec *codec) { @@ -13448,7 +13512,7 @@ static void alc269_speaker_automute(struct hda_codec *codec) } /* unsolicited event for HP jack sensing */ -static void alc269_eeepc_unsol_event(struct hda_codec *codec, +static void alc269_laptop_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { @@ -13461,7 +13525,7 @@ static void alc269_eeepc_unsol_event(struct hda_codec *codec, } } -static void alc269_eeepc_dmic_setup(struct hda_codec *codec) +static void alc269_laptop_dmic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->ext_mic.pin = 0x18; @@ -13471,7 +13535,17 @@ static void alc269_eeepc_dmic_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc269_eeepc_amic_setup(struct hda_codec *codec) +static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 6; + spec->auto_mic = 1; +} + +static void alc269_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->ext_mic.pin = 0x18; @@ -13481,7 +13555,7 @@ static void alc269_eeepc_amic_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc269_eeepc_inithook(struct hda_codec *codec) +static void alc269_laptop_inithook(struct hda_codec *codec) { alc269_speaker_automute(codec); alc_mic_automute(codec); @@ -13494,22 +13568,10 @@ static struct hda_verb alc269_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the - * analog-loopback mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* - * Set up output mixers (0x0c - 0x0e) + * Set up output mixers (0x02 - 0x03) */ /* set vol=0 to output mixers */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -13534,26 +13596,57 @@ static struct hda_verb alc269_init_verbs[] = { {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* FIXME: use Mux-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x23, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* FIXME: use matrix-type input source selection */ + /* set EAPD */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + +static struct hda_verb alc269vb_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* + * Set up output mixers (0x02 - 0x03) + */ + /* set vol=0 to output mixers */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* FIXME: use Mux-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x00}, /* set EAPD */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; @@ -13601,6 +13694,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; + hda_nid_t real_capsrc_nids; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc269_ignore); @@ -13622,11 +13716,20 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - add_verb(spec, alc269_init_verbs); + if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { + add_verb(spec, alc269vb_init_verbs); + real_capsrc_nids = alc269vb_capsrc_nids[0]; + alc_ssid_check(codec, 0x21, 0x1b, 0x14); + } else { + add_verb(spec, alc269_init_verbs); + real_capsrc_nids = alc269_capsrc_nids[0]; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + } + spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; /* set default input source */ - snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], + snd_hda_codec_write_cache(codec, real_capsrc_nids, 0, AC_VERB_SET_CONNECT_SEL, spec->input_mux->items[0].index); @@ -13637,8 +13740,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(codec); - alc_ssid_check(codec, 0x15, 0x1b, 0x14); - return 1; } @@ -13664,8 +13765,8 @@ static void alc269_auto_init(struct hda_codec *codec) static const char *alc269_models[ALC269_MODEL_LAST] = { [ALC269_BASIC] = "basic", [ALC269_QUANTA_FL1] = "quanta", - [ALC269_ASUS_AMIC] = "asus-amic", - [ALC269_ASUS_DMIC] = "asus-dmic", + [ALC269_AMIC] = "laptop-amic", + [ALC269_DMIC] = "laptop-dmic", [ALC269_FUJITSU] = "fujitsu", [ALC269_LIFEBOOK] = "lifebook", [ALC269_AUTO] = "auto", @@ -13674,41 +13775,49 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { static struct snd_pci_quirk alc269_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1), SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", - ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80JT", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_DMIC), - SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_AMIC), - SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_AMIC), + ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC), + SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC), + SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC), + SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901", - ALC269_ASUS_DMIC), + ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101", - ALC269_ASUS_DMIC), - SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_ASUS_DMIC), - SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_ASUS_DMIC), + ALC269_DMIC), + SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), + SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), {} @@ -13738,47 +13847,75 @@ static struct alc_config_preset alc269_presets[] = { .setup = alc269_quanta_fl1_setup, .init_hook = alc269_quanta_fl1_init_hook, }, - [ALC269_ASUS_AMIC] = { - .mixers = { alc269_eeepc_mixer }, - .cap_mixer = alc269_epc_capture_mixer, + [ALC269_AMIC] = { + .mixers = { alc269_laptop_mixer }, + .cap_mixer = alc269_laptop_analog_capture_mixer, .init_verbs = { alc269_init_verbs, - alc269_eeepc_amic_init_verbs }, + alc269_laptop_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_eeepc_unsol_event, - .setup = alc269_eeepc_amic_setup, - .init_hook = alc269_eeepc_inithook, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_amic_setup, + .init_hook = alc269_laptop_inithook, }, - [ALC269_ASUS_DMIC] = { - .mixers = { alc269_eeepc_mixer }, - .cap_mixer = alc269_epc_capture_mixer, + [ALC269_DMIC] = { + .mixers = { alc269_laptop_mixer }, + .cap_mixer = alc269_laptop_digital_capture_mixer, .init_verbs = { alc269_init_verbs, - alc269_eeepc_dmic_init_verbs }, + alc269_laptop_dmic_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_eeepc_unsol_event, - .setup = alc269_eeepc_dmic_setup, - .init_hook = alc269_eeepc_inithook, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_dmic_setup, + .init_hook = alc269_laptop_inithook, + }, + [ALC269VB_AMIC] = { + .mixers = { alc269vb_laptop_mixer }, + .cap_mixer = alc269vb_laptop_analog_capture_mixer, + .init_verbs = { alc269vb_init_verbs, + alc269vb_laptop_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_amic_setup, + .init_hook = alc269_laptop_inithook, + }, + [ALC269VB_DMIC] = { + .mixers = { alc269vb_laptop_mixer }, + .cap_mixer = alc269vb_laptop_digital_capture_mixer, + .init_verbs = { alc269vb_init_verbs, + alc269vb_laptop_dmic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269vb_laptop_dmic_setup, + .init_hook = alc269_laptop_inithook, }, [ALC269_FUJITSU] = { .mixers = { alc269_fujitsu_mixer }, - .cap_mixer = alc269_epc_capture_mixer, + .cap_mixer = alc269_laptop_digital_capture_mixer, .init_verbs = { alc269_init_verbs, - alc269_eeepc_dmic_init_verbs }, + alc269_laptop_dmic_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_eeepc_unsol_event, - .setup = alc269_eeepc_dmic_setup, - .init_hook = alc269_eeepc_inithook, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_dmic_setup, + .init_hook = alc269_laptop_inithook, }, [ALC269_LIFEBOOK] = { .mixers = { alc269_lifebook_mixer }, @@ -13799,6 +13936,7 @@ static int patch_alc269(struct hda_codec *codec) struct alc_spec *spec; int board_config; int err; + int is_alc269vb = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -13815,6 +13953,7 @@ static int patch_alc269(struct hda_codec *codec) alc_free(codec); return -ENOMEM; } + is_alc269vb = 1; } board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, @@ -13850,7 +13989,7 @@ static int patch_alc269(struct hda_codec *codec) if (board_config != ALC269_AUTO) setup_preset(codec, &alc269_presets[board_config]); - if (codec->subsystem_id == 0x17aa3bf8) { + if (board_config == ALC269_QUANTA_FL1) { /* Due to a hardware problem on Lenovo Ideadpad, we need to * fix the sample rate of analog I/O to 44.1kHz */ @@ -13863,9 +14002,16 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; - spec->adc_nids = alc269_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); - spec->capsrc_nids = alc269_capsrc_nids; + if (!is_alc269vb) { + spec->adc_nids = alc269_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); + spec->capsrc_nids = alc269_capsrc_nids; + } else { + spec->adc_nids = alc269vb_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); + spec->capsrc_nids = alc269vb_capsrc_nids; + } + if (!spec->cap_mixer) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); -- cgit v0.10.2 From cec27c891b805b2ab2302f9fcbdacb6f179ac0d4 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 4 Feb 2010 14:18:18 +0100 Subject: ALSA: hda - Add support of ALC665 - Add support for ALC665 - Add more ASUS model - Modify common patch for ALC272 ALC273 ALC661 ALC662 ALC663 ALC665 Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 826ecdb..82772f0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16597,13 +16597,6 @@ static struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -16653,6 +16646,28 @@ static struct hda_verb alc662_init_verbs[] = { { } }; +static struct hda_verb alc663_init_verbs[] = { + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } +}; + +static struct hda_verb alc272_init_verbs[] = { + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } +}; + static struct hda_verb alc662_sue_init_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, @@ -16672,61 +16687,6 @@ static struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = { {} }; -/* - * generic initialization of ADC, input mixers and output mixers - */ -static struct hda_verb alc662_auto_init_verbs[] = { - /* - * Unmute ADC and set the default input to mic-in - */ - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for front - * panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* - * Set up output mixers (0x0c - 0x0f) - */ - /* set vol=0 to output mixers */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { } -}; - -/* additional verbs for ALC663 */ -static struct hda_verb alc663_auto_init_verbs[] = { - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - { } -}; - static struct hda_verb alc663_m51va_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -17477,6 +17437,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1), SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), @@ -17512,6 +17473,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), @@ -18157,9 +18119,13 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - add_verb(spec, alc662_auto_init_verbs); - if (codec->vendor_id == 0x10ec0663) - add_verb(spec, alc663_auto_init_verbs); + add_verb(spec, alc662_init_verbs); + if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || + codec->vendor_id == 0x10ec0665) + add_verb(spec, alc663_init_verbs); + + if (codec->vendor_id == 0x10ec0272) + add_verb(spec, alc272_init_verbs); err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -18251,11 +18217,20 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(codec); - if (codec->vendor_id == 0x10ec0662) + + switch (codec->vendor_id) { + case 0x10ec0662: set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - else + break; + case 0x10ec0272: + case 0x10ec0663: + case 0x10ec0665: set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); - + break; + case 0x10ec0273: + set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT); + break; + } spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; @@ -18305,6 +18280,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, + { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, -- cgit v0.10.2 From 350a514787a4516746f738f69bff6aa0d4ac70e9 Mon Sep 17 00:00:00 2001 From: Sebastien Alaiwan Date: Fri, 5 Feb 2010 08:58:20 +0100 Subject: ALSA: ice1712: fix: lock samplerate when samplerate locking is enabled I found that the sampling rate locking setting of the ice1712 sound driver was only half-respected : when the driver was locked to, let's say, 44100Hz, and a usermode app was requesting 48000Hz playback, the request was succesful although the soundcard would continue to run at 44100Hz. Here's a patch that will make those requests to fail. Signed-off-by: Sebastien Alaiwan Signed-off-by: Takashi Iwai diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index c7cff6f..fb61943 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1180,6 +1180,10 @@ static int snd_ice1712_playback_pro_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (is_pro_rate_locked(ice)) { + runtime->hw.rate_min = PRO_RATE_DEFAULT; + runtime->hw.rate_max = PRO_RATE_DEFAULT; + } if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); @@ -1197,6 +1201,11 @@ static int snd_ice1712_capture_pro_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (is_pro_rate_locked(ice)) { + runtime->hw.rate_min = PRO_RATE_DEFAULT; + runtime->hw.rate_max = PRO_RATE_DEFAULT; + } + return 0; } -- cgit v0.10.2 From c50749de02f272be6e09b9016e13a17307d29066 Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Fri, 5 Feb 2010 16:29:53 +0200 Subject: ASoC: pandora: Add DAC regulator support Pandora's external DAC is connected to VSIM TWL4030 supply, so let's start switching it too to save more power. Also DAC got it's own DAPM handler. Signed-off-by: Grazvydas Ignotas Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 68980c1..de10f76 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include @@ -40,6 +41,8 @@ #define PREFIX "ASoC omap3pandora: " +static struct regulator *omap3pandora_dac_reg; + static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, unsigned int fmt) { @@ -106,21 +109,37 @@ static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_CBS_CFS); } -static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, +static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { + /* + * The PCM1773 DAC datasheet requires 1ms delay between switching + * VCC power on/off and /PD pin high/low + */ if (SND_SOC_DAPM_EVENT_ON(event)) { + regulator_enable(omap3pandora_dac_reg); + mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); - gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); } else { - gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); - mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + mdelay(1); + regulator_disable(omap3pandora_dac_reg); } return 0; } +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); + else + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + + return 0; +} + /* * Audio paths on Pandora board: * @@ -130,7 +149,9 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM, + 0, 0, omap3pandora_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -306,8 +327,18 @@ static int __init omap3pandora_soc_init(void) goto fail2; } + omap3pandora_dac_reg = regulator_get(&omap3pandora_snd_device->dev, "vcc"); + if (IS_ERR(omap3pandora_dac_reg)) { + pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n", + dev_name(&omap3pandora_snd_device->dev), + PTR_ERR(omap3pandora_dac_reg)); + goto fail3; + } + return 0; +fail3: + platform_device_del(omap3pandora_snd_device); fail2: platform_device_put(omap3pandora_snd_device); fail1: @@ -320,6 +351,7 @@ module_init(omap3pandora_soc_init); static void __exit omap3pandora_soc_exit(void) { + regulator_put(omap3pandora_dac_reg); platform_device_unregister(omap3pandora_snd_device); gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); -- cgit v0.10.2 From 07f804495cb08c8fdf16eee8f7d90edce4a3c9c5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Feb 2010 15:06:13 +0100 Subject: ALSA: hda - Detect HP mute-LED GPIO setup from GPIO counts The GPIO pin number for the mute LED control on HP laptops can be determined more easily by checking the number of available GPIO pins of the codec chip. On a small package with up to 3 GPIOs, GPIO 0 is used while GPIO 3 is used for others. This fixes the missing mute GPIO for some HP laptops with new codecs. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 834c598..3996187 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4754,19 +4754,14 @@ static int hp_blike_system(u32 subsystem_id); static void set_hp_led_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - switch (codec->vendor_id) { - case 0x111d7608: - /* GPIO 0 */ - spec->gpio_led = 0x01; - break; - case 0x111d7600: - case 0x111d7601: - case 0x111d7602: - case 0x111d7603: - /* GPIO 3 */ - spec->gpio_led = 0x08; - break; - } + unsigned int gpio; + + gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + gpio &= AC_GPIO_IO_COUNT; + if (gpio > 3) + spec->gpio_led = 0x08; /* GPIO 3 */ + else + spec->gpio_led = 0x01; /* GPIO 0 */ } /* -- cgit v0.10.2 From c21bd0254371c207636e84c9e033d13a6fe48d43 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Feb 2010 15:16:08 +0100 Subject: ALSA: hda - Merge HP mute-LED status callback on both IDT 92HD7x and 8x codecs Merge the mute-LED status callback function for both IDT 92HD7x and 8x codecs to one function. Also it's changed to check all DACs, and called in the initialization to sync with the current status. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3996187..ea25423 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4363,6 +4363,12 @@ static int stac92xx_init(struct hda_codec *codec) if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) stac_issue_unsol_event(codec, nid); } + +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* sync mute LED */ + if (spec->gpio_led && codec->patch_ops.check_power_status) + codec->patch_ops.check_power_status(codec, 0x01); +#endif if (spec->dac_list) stac92xx_power_down(codec); return 0; @@ -4909,6 +4915,11 @@ static int stac92xx_resume(struct hda_codec *codec) stac_issue_unsol_event(codec, spec->autocfg.line_out_pins[0]); } +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* sync mute LED */ + if (spec->gpio_led && codec->patch_ops.check_power_status) + codec->patch_ops.check_power_status(codec, 0x01); +#endif return 0; } @@ -4928,43 +4939,29 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + int i, muted = 1; - if (nid == 0x10) { - if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & - HDA_AMP_MUTE) - spec->gpio_data &= ~spec->gpio_led; /* orange */ - else - spec->gpio_data |= spec->gpio_led; /* white */ - - if (!spec->gpio_led_polarity) { - /* LED state is inverted on these systems */ - spec->gpio_data ^= spec->gpio_led; + for (i = 0; i < spec->multiout.num_dacs; i++) { + nid = spec->multiout.dac_nids[i]; + if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE)) { + muted = 0; /* something heard */ + break; } - - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, - spec->gpio_data); } + if (muted) + spec->gpio_data &= ~spec->gpio_led; /* orange */ + else + spec->gpio_data |= spec->gpio_led; /* white */ - return 0; -} - -static int idt92hd83xxx_hp_check_power_status(struct hda_codec *codec, - hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; + if (!spec->gpio_led_polarity) { + /* LED state is inverted on these systems */ + spec->gpio_data ^= spec->gpio_led; + } - if (nid != 0x13) - return 0; - if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data |= spec->gpio_led; /* mute LED on */ - else - spec->gpio_data &= ~spec->gpio_led; /* mute LED off */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); - return 0; } - #endif static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) @@ -5361,7 +5358,7 @@ again: spec->gpio_data |= spec->gpio_led; /* register check_power_status callback. */ codec->patch_ops.check_power_status = - idt92hd83xxx_hp_check_power_status; + stac92xx_hp_check_power_status; } #endif -- cgit v0.10.2 From b99a776d0b17ae0f3a54e86009887a00ac4889d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Feb 2010 15:21:09 +0100 Subject: ALSA: hda - Remove static gpio_led setup via model We have now a better mute-LED GPIO detection, and no need to assign the values statically per model option. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea25423..ec0637e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5343,9 +5343,6 @@ again: codec->patch_ops = stac92xx_patch_ops; - if (spec->board_config == STAC_92HD83XXX_HP) - spec->gpio_led = 0x01; - if (find_mute_led_gpio(codec)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, @@ -5673,7 +5670,6 @@ again: */ spec->num_smuxes = 1; spec->num_dmuxes = 1; - spec->gpio_led = 0x01; /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); @@ -5688,8 +5684,6 @@ again: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; - /* orange/white mute led on GPIO3, orange=0, white=1 */ - spec->gpio_led = 0x08; break; } -- cgit v0.10.2 From dce17d4ff366230aeeaaf42512bba3711243cf1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Feb 2010 09:25:26 +0100 Subject: ALSA: hda - Fix default polarity of mute-LED GPIO on 92HD83x/88x codecs The previous commit caused a regression on HP laptops with 92HD83x/88x codecs. The default polarity of mute-LED GPIO is inverted on these devices. Reference: Novell bnc#578190 https://bugzilla.novell.com/show_bug.cgi?id=578190 Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ec0637e..8c416bb 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4790,7 +4790,7 @@ static void set_hp_led_gpio(struct hda_codec *codec) * Need more information on whether it is true across the entire series. * -- kunal */ -static int find_mute_led_gpio(struct hda_codec *codec) +static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity) { struct sigmatel_spec *spec = codec->spec; const struct dmi_device *dev = NULL; @@ -4817,7 +4817,7 @@ static int find_mute_led_gpio(struct hda_codec *codec) */ if (!hp_blike_system(codec->subsystem_id)) { set_hp_led_gpio(codec); - spec->gpio_led_polarity = 1; + spec->gpio_led_polarity = default_polarity; return 1; } } @@ -5343,7 +5343,7 @@ again: codec->patch_ops = stac92xx_patch_ops; - if (find_mute_led_gpio(codec)) + if (find_mute_led_gpio(codec, 0)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); @@ -5705,7 +5705,7 @@ again: } } - if (find_mute_led_gpio(codec)) + if (find_mute_led_gpio(codec, 1)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); -- cgit v0.10.2 From cebe41d4b8f8092359de31e241815fcb4b4dc0be Mon Sep 17 00:00:00 2001 From: Alexey Dobriyan Date: Sat, 6 Feb 2010 00:21:03 +0200 Subject: sound: use DEFINE_PCI_DEVICE_TABLE Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to .devinit.rodata section, so they can be discarded in some cases, and make them const. Signed-off-by: Alexey Dobriyan Signed-off-by: Takashi Iwai diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 89466b0..24d152c 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -198,7 +198,7 @@ MODULE_LICENSE("GPL"); * 5530 only. The 5510/5520 decode is different. */ -static struct pci_device_id id_tbl[] = { +static DEFINE_PCI_DEVICE_TABLE(id_tbl) = { { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 8f5098f..4382d0f 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1048,7 +1048,7 @@ snd_ad1889_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_device_id snd_ad1889_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_ANALOG_DEVICES, PCI_DEVICE_ID_AD1889JS) }, { 0, }, }; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index aaf4da6..5c6e322 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -275,7 +275,7 @@ struct snd_ali { #endif }; -static struct pci_device_id snd_ali_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ali_ids) = { {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0}, {0, } }; diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 3aa35af..d7653cb 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -145,7 +145,7 @@ struct snd_als300_substream_data { int block_counter_register; }; -static struct pci_device_id snd_als300_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_als300_ids) = { { 0x4005, 0x0300, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300 }, { 0x4005, 0x0308, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300_PLUS }, { 0, } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 3dbacde..d75cf7b 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -117,7 +117,7 @@ struct snd_card_als4000 { #endif }; -static struct pci_device_id snd_als4000_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_als4000_ids) = { { 0x4005, 0x4000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ALS4000 */ { 0, } }; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752df..81e2bfc 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -286,7 +286,7 @@ struct atiixp { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index e7e147b..91d7036 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -261,7 +261,7 @@ struct atiixp_modem { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index c0e8c6b..aa51cc7 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,6 +1,6 @@ #include "au8810.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index a652733..2f321e7 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,6 +1,6 @@ #include "au8820.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index 6c702ad..279b78f 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,6 +1,6 @@ #include "au8830.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 4d34bb0..67921f9 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -164,7 +164,7 @@ MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); -static struct pci_device_id snd_aw2_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, 0, 0, 0}, {0} diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 69867ac..4679ed8 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -350,7 +350,7 @@ struct snd_azf3328 { #endif }; -static const struct pci_device_id snd_azf3328_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_azf3328_ids) = { { 0x122D, 0x50DC, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* PCI168/3328 */ { 0x122D, 0x80DA, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* 3328 */ { 0, } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 4e2b925..37e1b5d 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -795,7 +795,7 @@ fail: .driver_data = SND_BT87X_BOARD_ ## id } /* driver_data is the card id for that device */ -static struct pci_device_id snd_bt87x_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_ids) = { /* Hauppauge WinTV series */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC), /* Hauppauge WinTV series */ @@ -964,7 +964,7 @@ static void __devexit snd_bt87x_remove(struct pci_dev *pci) /* default entries for all Bt87x cards - it's not exported */ /* driver_data is set to 0 to call detection */ -static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), { } diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 15e4138..0a3d3d6 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1875,7 +1875,7 @@ static int snd_ca0106_resume(struct pci_dev *pci) #endif // PCI IDs -static struct pci_device_id snd_ca0106_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = { { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */ { 0, } }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index a312bae..1ded64e 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2796,7 +2796,7 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} #endif -static struct pci_device_id snd_cmipci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cmipci_ids) = { {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, @@ -3018,7 +3018,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc int integrated_midi = 0; char modelstr[16]; int pcm_index, pcm_spdif_index; - static struct pci_device_id intel_82437vx[] = { + static DEFINE_PCI_DEVICE_TABLE(intel_82437vx) = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) }, { }, }; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index e2e0359..9edc650 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -494,7 +494,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_cs4281_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs4281_ids) = { { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 033aec4..767fa7f 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -64,7 +64,7 @@ MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); -static struct pci_device_id snd_cs46xx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs46xx_ids) = { { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index dc46432..207479a 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -58,7 +58,7 @@ struct snd_cs5530 { unsigned long pci_base; }; -static struct pci_device_id snd_cs5530_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs5530_ids) = { {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0}, {0,} diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 91e7faf..afb8037 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); -static struct pci_device_id snd_cs5535audio_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs5535audio_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, {} diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index ed44ed7..f42e7e1 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -43,7 +43,7 @@ MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); module_param_array(subsystem, int, NULL, 0444); MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); -static struct pci_device_id ct_pci_dev_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(ct_pci_dev_ids) = { /* only X-Fi is supported, so... */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1), .driver_data = ATC20K1, diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index 8c6db3a..a65bafe 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -63,7 +63,7 @@ static const struct firmware card_fw[] = { {0, "darla20_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0}, /* DSP 56301 Darla20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 04cbf3e..0a6c50b 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "darla24_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0}, /* DSP 56301 Darla24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0}, /* DSP 56301 Darla24 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index 4022e43..f514279 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -81,7 +81,7 @@ static const struct firmware card_fw[] = { {0, "3g_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0}, /* Echo 3G */ {0,} }; diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index c0e64b8..2364f8a 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "gina20_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0}, /* DSP 56301 Gina20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index c36a78d..616b558 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -85,7 +85,7 @@ static const struct firmware card_fw[] = { {0, "gina24_361_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56301 Gina24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56301 Gina24 rev.1 */ {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56361 Gina24 rev.0 */ diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index 0a58a7c..776175c 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0}, /* Indigo */ {0,} }; diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index 2db24d2..8816b0b 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dj_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/ {0,} }; diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index 2e44316..b1e3652 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_djx_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index a60c0a0..1035125 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_io_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index eb3819f..60b7cb2 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_iox_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ {0,} }; diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 5061946..8c3f5c5 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -76,7 +76,7 @@ static const struct firmware card_fw[] = { {0, "layla20_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0}, /* DSP 56301 Layla20 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0}, /* DSP 56301 Layla20 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index e09e3ea..ed1cc0a 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -87,7 +87,7 @@ static const struct firmware card_fw[] = { {0, "layla24_2S_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0}, /* DSP 56361 Layla24 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index f05c8c0..cc2bbfc 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -77,7 +77,7 @@ static const struct firmware card_fw[] = { {0, "mia_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0}, /* DSP 56361 Mia rev.0 */ {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0}, /* DSP 56361 Mia rev.1 */ {0,} diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index b05bad9..3e7e018 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -92,7 +92,7 @@ static const struct firmware card_fw[] = { {0, "mona_2_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56301 Mona rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56301 Mona rev.1 */ {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56301 Mona rev.2 */ diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 168af67..4203782 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -76,7 +76,7 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ -static struct pci_device_id snd_emu10k1_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1_ids) = { { PCI_VDEVICE(CREATIVE, 0x0002), 0 }, /* EMU10K1 */ { PCI_VDEVICE(CREATIVE, 0x0004), 1 }, /* Audigy */ { PCI_VDEVICE(CREATIVE, 0x0008), 1 }, /* Audigy 2 Value SB0400 */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 1d369ff..df47f73 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1605,7 +1605,7 @@ static void __devexit snd_emu10k1x_remove(struct pci_dev *pci) } // PCI IDs -static struct pci_device_id snd_emu10k1x_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = { { PCI_VDEVICE(CREATIVE, 0x0006), 0 }, /* Dell OEM version (EMU10K1) */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 2b82c5c..c7fba53 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -443,7 +443,7 @@ struct ensoniq { static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_audiopci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_audiopci_ids) = { #ifdef CHIP1370 { PCI_VDEVICE(ENSONIQ, 0x5000), 0, }, /* ES1370 */ #endif diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fb83e1f..553b752 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -243,7 +243,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_es1938_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_es1938_ids) = { { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a11f453..ecaea9f 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -551,7 +551,7 @@ struct es1968 { static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_es1968_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_es1968_ids) = { /* Maestro 1 */ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO }, /* Maestro 2 */ diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 83508b3..e1baad7 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -205,7 +205,7 @@ struct fm801 { #endif }; -static struct pci_device_id snd_fm801_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_fm801_ids) = { { 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */ { 0, } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1f516e6..ac05bef 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2664,7 +2664,7 @@ static void __devexit azx_remove(struct pci_dev *pci) } /* PCI IDs */ -static struct pci_device_id azx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* ICH 6..10 */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index fb61943..4fc6d8b 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -106,7 +106,7 @@ module_param_array(dxr_enable, int, NULL, 0444); MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); -static const struct pci_device_id snd_ice1712_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ice1712_ids) = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ae29073..c1498fa 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -94,7 +94,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ -static const struct pci_device_id snd_vt1724_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vt1724_ids) = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index b990143..6433e65 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -420,7 +420,7 @@ struct intel8x0 { u32 int_sta_mask; /* interrupt status mask */ }; -static struct pci_device_id snd_intel8x0_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0_ids) = { { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 9e7d12e..13cec1e 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -219,7 +219,7 @@ struct intel8x0m { unsigned int pcm_pos_shift; }; -static struct pci_device_id snd_intel8x0m_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 7cc38a1..6d79570 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -418,7 +418,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); MODULE_AUTHOR("Haroldo Gamal "); -static struct pci_device_id snd_korg1212_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_korg1212_ids) = { { .vendor = 0x10b5, .device = 0x906d, diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 11b8c65..0cca560 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -55,7 +55,7 @@ static const char card_name[] = "LX6464ES"; #define PCI_DEVICE_ID_PLX_LX6464ES PCI_DEVICE_ID_PLX_9056 -static struct pci_device_id snd_lx6464es_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_lx6464es_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES), .subvendor = PCI_VENDOR_ID_DIGIGRAM, .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 75283fbb..b64e781 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -861,7 +861,7 @@ struct snd_m3 { /* * pci ids */ -static struct pci_device_id snd_m3_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = { {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO_1, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO, PCI_ANY_ID, PCI_ANY_ID, diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index a83d196..7e8e7da 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -60,7 +60,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); /* */ -static struct pci_device_id snd_mixart_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_mixart_ids) = { { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 97a0731..5a60492 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -262,7 +262,7 @@ struct nm256 { /* * PCI ids */ -static struct pci_device_id snd_nm256_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_nm256_ids) = { {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index e3c229b..5a87d68 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -48,7 +48,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static struct pci_device_id hifier_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(hifier_ids) = { { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index acbedeb..289cb4d 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -72,7 +72,7 @@ enum { MODEL_CLARO_HALO, /* HT-Omega Claro halo */ }; -static struct pci_device_id oxygen_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0218), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF }, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 563b6f5..f03a2f2 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -40,7 +40,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static struct pci_device_id xonar_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, { OXYGEN_PCI_SUBID(0x1043, 0x8275) }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 833e9c7..95cfde2 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -94,7 +94,7 @@ enum { PCI_ID_LAST }; -static struct pci_device_id pcxhr_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b5ca02e..bb08a28 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -506,7 +506,7 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip); /* */ -static struct pci_device_id snd_riptide_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_riptide_ids) = { { PCI_DEVICE(0x127a, 0x4310) }, { PCI_DEVICE(0x127a, 0x4320) }, { PCI_DEVICE(0x127a, 0x4330) }, @@ -515,7 +515,7 @@ static struct pci_device_id snd_riptide_ids[] = { }; #ifdef SUPPORT_JOYSTICK -static struct pci_device_id snd_riptide_joystick_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_riptide_joystick_ids) = { { PCI_DEVICE(0x127a, 0x4312) }, { PCI_DEVICE(0x127a, 0x4322) }, { PCI_DEVICE(0x127a, 0x4332) }, diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index f977dba..d5e1c6e 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -226,7 +226,7 @@ struct rme32 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme32_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme32_ids) = { {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 2ba5c0f..9d5252b 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -231,7 +231,7 @@ struct rme96 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme96_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme96_ids) = { { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7bb827c..52c6eb5 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -585,7 +585,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_hdsp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_hdsp_ids) = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a1b10d1..3d72c1e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -512,7 +512,7 @@ static char channel_map_madi_ss[HDSPM_MAX_CHANNELS] = { }; -static struct pci_device_id snd_hdspm_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_hdspm_ids) = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI, diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index bc539ab..44a3e2d 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -314,7 +314,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_rme9652_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme9652_ids) = { { .vendor = 0x10ee, .device = 0x3fc4, diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 1a5ff06..7e3e8fb 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -48,7 +48,7 @@ MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); -static struct pci_device_id snd_sis7019_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, { 0, } }; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 1f6406c..337b9fa 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -242,7 +242,7 @@ struct sonicvibes { #endif }; -static struct pci_device_id snd_sonic_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_sonic_ids) = { { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 21cef97..6d05818 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -62,7 +62,7 @@ MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM."); module_param_array(wavetable_size, int, NULL, 0444); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); -static struct pci_device_id snd_trident_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_trident_ids) = { {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8a332d2..9595b5b 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -401,7 +401,7 @@ struct via82xx { #endif }; -static struct pci_device_id snd_via82xx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_ids) = { /* 0x1106, 0x3058 */ { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 47eb615..f7e8bbbe 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -260,7 +260,7 @@ struct via82xx_modem { struct snd_info_entry *proc_entry; }; -static struct pci_device_id snd_via82xx_modem_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_modem_ids) = { { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index fc9136c..99a9a81 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -60,7 +60,7 @@ enum { VX_PCI_VX222_NEW }; -static struct pci_device_id snd_vx222_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vx222_ids) = { { 0x10b5, 0x9050, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_OLD, }, /* PLX */ { 0x10b5, 0x9030, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_NEW, }, /* PLX */ { 0, } diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e6b18b9..80c6821 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address"); module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); -static struct pci_device_id snd_ymfpci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ymfpci_ids) = { { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */ -- cgit v0.10.2 From 22313eafe92aeec1db9839f5afb71675bf2a5c33 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Feb 2010 10:42:33 +0000 Subject: ASoC: add phycore-ac97 sound support This patch adds sound support for Phytec PhyCORE / PhyCARD modules in AC97 mode. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index d05cc95..9f8bb92 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -8,3 +8,5 @@ endif obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support +snd-soc-phycore-ac97-objs := phycore-ac97.o +obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c new file mode 100644 index 0000000..a8307d5 --- /dev/null +++ b/sound/soc/imx/phycore-ac97.c @@ -0,0 +1,90 @@ +/* + * phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode + * + * Copyright 2009 Sascha Hauer, Pengutronix + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/wm9712.h" +#include "imx-ssi.h" + +static struct snd_soc_card imx_phycore; + +static struct snd_soc_ops imx_phycore_hifi_ops = { +}; + +static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { + { + .name = "HiFi", + .stream_name = "HiFi", + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .ops = &imx_phycore_hifi_ops, + }, +}; + +static struct snd_soc_card imx_phycore = { + .name = "PhyCORE-audio", + .platform = &imx_soc_platform, + .dai_link = imx_phycore_dai_ac97, + .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), +}; + +static struct snd_soc_device imx_phycore_snd_devdata = { + .card = &imx_phycore, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *imx_phycore_snd_device; + +static int __init imx_phycore_init(void) +{ + int ret; + + if (!machine_is_pcm043() && !machine_is_pca100()) + /* return happy. We might run on a totally different machine */ + return 0; + + imx_phycore_snd_device = platform_device_alloc("soc-audio", -1); + if (!imx_phycore_snd_device) + return -ENOMEM; + + imx_phycore_dai_ac97[0].cpu_dai = &imx_ssi_pcm_dai[0]; + + platform_set_drvdata(imx_phycore_snd_device, &imx_phycore_snd_devdata); + imx_phycore_snd_devdata.dev = &imx_phycore_snd_device->dev; + ret = platform_device_add(imx_phycore_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(imx_phycore_snd_device); + } + + return ret; +} + +static void __exit imx_phycore_exit(void) +{ + platform_device_unregister(imx_phycore_snd_device); +} + +late_initcall(imx_phycore_init); +module_exit(imx_phycore_exit); + +MODULE_AUTHOR("Sascha Hauer "); +MODULE_DESCRIPTION("PhyCORE ALSA SoC driver"); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From c0ff4bcd2e8505b09e0bedc74d08ad2da1e326f8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 9 Feb 2010 02:32:59 +0800 Subject: ASoC: cs4270: enable regulators at probe time Enable the bulk regulators at probe time so we can safely disable them again when going to suspend without confusing the reference counter. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 593bfc7..dfbeb2d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -629,8 +629,17 @@ static int cs4270_probe(struct platform_device *pdev) if (ret < 0) goto error_free_pcms; + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret < 0) + goto error_free_regulators; + return 0; +error_free_regulators: + regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + error_free_pcms: snd_soc_free_pcms(socdev); @@ -650,6 +659,7 @@ static int cs4270_remove(struct platform_device *pdev) struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return 0; -- cgit v0.10.2 From c42a59ea277a8898b8f7c83fc89b00be225ea6aa Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Feb 2010 15:24:04 +0200 Subject: ASoC: TWL4030: Add supply for audio serial interface control The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b32aeb3..277862e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -55,7 +55,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x0c, /* REG_ATXR1PGA (0xB) */ 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ - 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_AUDIO_IF (0xE) */ 0x00, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ @@ -1203,6 +1203,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), + /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1337,6 +1339,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, @@ -1455,6 +1462,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "APLL Enable"}, {"ADC Virtual Right2", NULL, "APLL Enable"}, + {"ADC Virtual Left1", NULL, "AIF Enable"}, + {"ADC Virtual Right1", NULL, "AIF Enable"}, + {"ADC Virtual Left2", NULL, "AIF Enable"}, + {"ADC Virtual Right2", NULL, "AIF Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, -- cgit v0.10.2 From c6848bf566c7217a6090693ff5cc47091fa772f5 Mon Sep 17 00:00:00 2001 From: Paul Menzel Date: Tue, 9 Feb 2010 11:42:27 +0100 Subject: ASoC: Typo. s/Freecale/Freescale/ Signed-off-by: Paul Menzel Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 5f006f0d..c7d0fd9 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,5 +1,5 @@ config SND_IMX_SOC - tristate "SoC Audio for Freecale i.MX CPUs" + tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC && BROKEN select SND_PCM select FIQ -- cgit v0.10.2 From c3a3e040f01457d2ea4f199f75ca205401001a3b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 11 Feb 2010 17:50:44 +0100 Subject: ALSA: usbmixer - add possibility to remap dB values USB devices tends to represent dB ranges in different way than ALSA expects. Add possibility to override these values and add guessed values for SoundBlaster MP3+. Also rename 'Capture Input Source' control to 'Capture Source' for SoundBlaster MP3+ and Extigy. Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index c998220..c72ad0c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -123,6 +123,7 @@ struct usb_mixer_elem_info { int channels; int val_type; int min, max, res; + int dBmin, dBmax; int cached; int cache_val[MAX_CHANNELS]; u8 initialized; @@ -194,42 +195,50 @@ enum { */ #include "usbmixer_maps.c" -/* get the mapped name if the unit matches */ -static int check_mapped_name(struct mixer_build *state, int unitid, int control, char *buf, int buflen) +static const struct usbmix_name_map * +find_map(struct mixer_build *state, int unitid, int control) { - const struct usbmix_name_map *p; + const struct usbmix_name_map *p = state->map; - if (! state->map) - return 0; + if (!p) + return NULL; for (p = state->map; p->id; p++) { - if (p->id == unitid && p->name && - (! control || ! p->control || control == p->control)) { - buflen--; - return strlcpy(buf, p->name, buflen); - } + if (p->id == unitid && + (!control || !p->control || control == p->control)) + return p; } - return 0; + return NULL; } -/* check whether the control should be ignored */ -static int check_ignored_ctl(struct mixer_build *state, int unitid, int control) +/* get the mapped name if the unit matches */ +static int +check_mapped_name(const struct usbmix_name_map *p, char *buf, int buflen) { - const struct usbmix_name_map *p; + if (!p || !p->name) + return 0; - if (! state->map) + buflen--; + return strlcpy(buf, p->name, buflen); +} + +/* check whether the control should be ignored */ +static inline int +check_ignored_ctl(const struct usbmix_name_map *p) +{ + if (!p || p->name || p->dB) return 0; - for (p = state->map; p->id; p++) { - if (p->id == unitid && ! p->name && - (! control || ! p->control || control == p->control)) { - /* - printk(KERN_DEBUG "ignored control %d:%d\n", - unitid, control); - */ - return 1; - } + return 1; +} + +/* dB mapping */ +static inline void check_mapped_dB(const struct usbmix_name_map *p, + struct usb_mixer_elem_info *cval) +{ + if (p && p->dB) { + cval->dBmin = p->dB->min; + cval->dBmax = p->dB->max; } - return 0; } /* get the mapped selector source name */ @@ -466,20 +475,8 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, if (size < sizeof(scale)) return -ENOMEM; - /* USB descriptions contain the dB scale in 1/256 dB unit - * while ALSA TLV contains in 1/100 dB unit - */ - scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256; - scale[3] = (convert_signed_value(cval, cval->max) * 100) / 256; - if (scale[3] <= scale[2]) { - /* something is wrong; assume it's either from/to 0dB */ - if (scale[2] < 0) - scale[3] = 0; - else if (scale[2] > 0) - scale[2] = 0; - else /* totally crap, return an error */ - return -EINVAL; - } + scale[2] = cval->dBmin; + scale[3] = cval->dBmax; if (copy_to_user(_tlv, scale, sizeof(scale))) return -EFAULT; return 0; @@ -720,6 +717,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) cval->min = default_min; cval->max = cval->min + 1; cval->res = 1; + cval->dBmin = cval->dBmax = 0; if (cval->val_type == USB_MIXER_BOOLEAN || cval->val_type == USB_MIXER_INV_BOOLEAN) { @@ -787,6 +785,24 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) cval->initialized = 1; } + + /* USB descriptions contain the dB scale in 1/256 dB unit + * while ALSA TLV contains in 1/100 dB unit + */ + cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 256; + cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 256; + if (cval->dBmin > cval->dBmax) { + /* something is wrong; assume it's either from/to 0dB */ + if (cval->dBmin < 0) + cval->dBmax = 0; + else if (cval->dBmin > 0) + cval->dBmin = 0; + if (cval->dBmin > cval->dBmax) { + /* totally crap, return an error */ + return -EINVAL; + } + } + return 0; } @@ -912,6 +928,7 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, int nameid = desc[desc[0] - 1]; struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + const struct usbmix_name_map *map; control++; /* change from zero-based to 1-based value */ @@ -920,7 +937,8 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, return; } - if (check_ignored_ctl(state, unitid, control)) + map = find_map(state, unitid, control); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -954,10 +972,11 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, } kctl->private_free = usb_mixer_elem_free; - len = check_mapped_name(state, unitid, control, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); mapped_name = len != 0; if (! len && nameid) - len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, sizeof(kctl->id.name)); switch (control) { case USB_FEATURE_MUTE: @@ -995,6 +1014,7 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + check_mapped_dB(map, cval); } break; @@ -1122,8 +1142,10 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, unsigned int num_outs = desc[5 + input_pins]; unsigned int i, len; struct snd_kcontrol *kctl; + const struct usbmix_name_map *map; - if (check_ignored_ctl(state, unitid, 0)) + map = find_map(state, unitid, 0); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1152,7 +1174,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, } kctl->private_free = usb_mixer_elem_free; - len = check_mapped_name(state, unitid, 0, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (! len) len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0); if (! len) @@ -1342,6 +1364,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned int i, err, nameid, type, len; struct procunit_info *info; struct procunit_value_info *valinfo; + const struct usbmix_name_map *map; static struct procunit_value_info default_value_info[] = { { 0x01, "Switch", USB_MIXER_BOOLEAN }, { 0 } @@ -1371,7 +1394,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned /* FIXME: bitmap might be longer than 8bit */ if (! (dsc[12 + num_ins] & (1 << (valinfo->control - 1)))) continue; - if (check_ignored_ctl(state, unitid, valinfo->control)) + map = find_map(state, unitid, valinfo->control); + if (check_ignored_ctl(map)) continue; cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (! cval) { @@ -1402,8 +1426,9 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned } kctl->private_free = usb_mixer_elem_free; - if (check_mapped_name(state, unitid, cval->control, kctl->id.name, sizeof(kctl->id.name))) - ; + if (check_mapped_name(map, kctl->id.name, + sizeof(kctl->id.name))) + /* nothing */ ; else if (info->name) strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name)); else { @@ -1542,6 +1567,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi int err; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; + const struct usbmix_name_map *map; char **namelist; if (! num_ins || desc[0] < 5 + num_ins) { @@ -1557,7 +1583,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi if (num_ins == 1) /* only one ? nonsense! */ return 0; - if (check_ignored_ctl(state, unitid, 0)) + map = find_map(state, unitid, 0); + if (check_ignored_ctl(map)) return 0; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1612,7 +1639,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi kctl->private_free = usb_mixer_selector_elem_free; nameid = desc[desc[0] - 1]; - len = check_mapped_name(state, unitid, 0, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (len) ; else if (nameid) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index 77c3588..79e903a 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -19,11 +19,16 @@ * */ +struct usbmix_dB_map { + u32 min; + u32 max; +}; struct usbmix_name_map { int id; const char *name; int control; + struct usbmix_dB_map *dB; }; struct usbmix_selector_map { @@ -72,7 +77,7 @@ static struct usbmix_name_map extigy_map[] = { { 8, "Line Playback" }, /* FU */ /* 9: IT mic */ { 10, "Mic Playback" }, /* FU */ - { 11, "Capture Input Source" }, /* SU */ + { 11, "Capture Source" }, /* SU */ { 12, "Capture" }, /* FU */ /* 13: OT pcm capture */ /* 14: MU (w/o controls) */ @@ -102,6 +107,9 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov */ +static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0}; /* just guess */ +static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */ + static struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ /* 2: IT mic */ @@ -110,16 +118,19 @@ static struct usbmix_name_map mp3plus_map[] = { /* 5: OT digital out */ /* 6: OT speaker */ /* 7: OT pcm capture */ - { 8, "Capture Input Source" }, /* FU, default PCM Capture Source */ + { 8, "Capture Source" }, /* FU, default PCM Capture Source */ /* (Mic, Input 1 = Line input, Input 2 = Optical input) */ { 9, "Master Playback" }, /* FU, default Speaker 1 */ /* { 10, "Mic Capture", 1 }, */ /* FU, Mic Capture */ - /* { 10, "Mic Capture", 2 }, */ /* FU, Mic Capture */ + { 10, /* "Mic Capture", */ NULL, 2, .dB = &mp3plus_dB_2 }, + /* FU, Mic Capture */ { 10, "Mic Boost", 7 }, /* FU, default Auto Gain Input */ - { 11, "Line Capture" }, /* FU, default PCM Capture */ + { 11, "Line Capture", .dB = &mp3plus_dB_2 }, + /* FU, default PCM Capture */ { 12, "Digital In Playback" }, /* FU, default PCM 1 */ - /* { 13, "Mic Playback" }, */ /* FU, default Mic Playback */ - { 14, "Line Playback" }, /* FU, default Speaker */ + { 13, /* "Mic Playback", */ .dB = &mp3plus_dB_1 }, + /* FU, default Mic Playback */ + { 14, "Line Playback", .dB = &mp3plus_dB_1 }, /* FU, default Speaker */ /* 15: MU */ { 0 } /* terminator */ }; -- cgit v0.10.2 From 867af973a3b38f2a564d612326efd2694d931f30 Mon Sep 17 00:00:00 2001 From: Thomas Weber Date: Thu, 11 Feb 2010 16:13:59 +0100 Subject: Add ASoC support for Devkit8000 This patch expands the omap3beagle sound soc for the beagle board clone DevKit8000. Signed-off-by: Thomas Weber Acked-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..18ebdc7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -94,12 +94,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA Say Y if you want to add support for SoC audio on the OMAP3 Pandora. config SND_OMAP_SOC_OMAP3_BEAGLE - tristate "SoC Audio support for OMAP3 Beagle" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle and Devkit8000" + depends on TWL4030_CORE && SND_OMAP_SOC + depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help - Say Y if you want to add support for SoC audio on the Beagleboard. + Say Y if you want to add support for SoC audio on the Beagleboard or + the clone Devkit8000. config SND_OMAP_SOC_ZOOM2 tristate "SoC Audio support for Zoom2" diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index d88ad5c..240e097 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -117,11 +117,11 @@ static int __init omap3beagle_soc_init(void) { int ret; - if (!machine_is_omap3_beagle()) { - pr_debug("Not OMAP3 Beagle!\n"); + if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) { + pr_debug("Not OMAP3 Beagle or Devkit8000!\n"); return -ENODEV; } - pr_info("OMAP3 Beagle SoC init\n"); + pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); if (!omap3beagle_snd_device) { -- cgit v0.10.2 From 6db29675b1cb60e878d04a1f69aba265189b2e33 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 11 Feb 2010 18:11:10 +0100 Subject: ASoC: fix compile breakage if CONFIG_SH_DMA_API=y && CONFIG_SND_SIU_MIGOR!=n Audio on Migo-R cannot work if CONFIG_SH_DMA_API=y, but compilation should not break anyway. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index a86696b..1066749 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_SH4_FSI config SND_SOC_SH4_SIU tristate depends on (SUPERH || ARCH_SHMOBILE) && HAVE_CLK + select DMA_ENGINE select DMADEVICES select SH_DMAE -- cgit v0.10.2 From 3a66d3877eaa4ab9818000a15c07326adaa9ca79 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 11 Feb 2010 13:27:19 +0000 Subject: ASoC: Add WM2000 driver MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The WM2000 is a low power, high quality handset receiver speaker driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It provides enhanced voice communication quality in a noisy environment if the handset acoustics are designed appropriately. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/wm2000.h b/include/sound/wm2000.h new file mode 100644 index 0000000..aa388ca --- /dev/null +++ b/include/sound/wm2000.h @@ -0,0 +1,26 @@ +/* + * linux/sound/wm2000.h -- Platform data for WM2000 + * + * Copyright 2010 Wolfson Microelectronics. PLC. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_WM2000_H +#define __LINUX_SND_WM2000_H + +struct wm2000_platform_data { + /** Filename for system-specific image to download to device. */ + const char *download_file; + + /** Divide MCLK by 2 for system clock? */ + unsigned int mclkdiv2:1; + + /** Disable speech clarity enhancement, for use when an + * external algorithm is used. */ + unsigned int speech_enh_disable:1; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5ab5921..1743d56 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C + select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI @@ -265,3 +266,6 @@ config SND_SOC_MAX9877 config SND_SOC_TPA6130A2 tristate + +config SND_SOC_WM2000 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 209dd6c..dd5ce6d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -58,6 +58,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-wm2000-objs := wm2000.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o @@ -119,3 +120,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c new file mode 100644 index 0000000..217b026 --- /dev/null +++ b/sound/soc/codecs/wm2000.c @@ -0,0 +1,888 @@ +/* + * wm2000.c -- WM2000 ALSA Soc Audio driver + * + * Copyright 2008-2010 Wolfson Microelectronics PLC. + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * The download image for the WM2000 will be requested as + * 'wm2000_anc.bin' by default (overridable via platform data) at + * runtime and is expected to be in flat binary format. This is + * generated by Wolfson configuration tools and includes + * system-specific callibration information. If supplied as a + * sequence of ASCII-encoded hexidecimal bytes this can be converted + * into a flat binary with a command such as this on the command line: + * + * perl -e 'while (<>) { s/[\r\n]+// ; printf("%c", hex($_)); }' + * < file > wm2000_anc.bin + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include + +#include "wm2000.h" + +enum wm2000_anc_mode { + ANC_ACTIVE = 0, + ANC_BYPASS = 1, + ANC_STANDBY = 2, + ANC_OFF = 3, +}; + +struct wm2000_priv { + struct i2c_client *i2c; + + enum wm2000_anc_mode anc_mode; + + unsigned int anc_active:1; + unsigned int anc_eng_ena:1; + unsigned int spk_ena:1; + + unsigned int mclk_div:1; + unsigned int speech_clarity:1; + + int anc_download_size; + char *anc_download; +}; + +static struct i2c_client *wm2000_i2c; + +static int wm2000_write(struct i2c_client *i2c, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + int ret; + + data[0] = (reg >> 8) & 0xff; + data[1] = reg & 0xff; + data[2] = value & 0xff; + + dev_vdbg(&i2c->dev, "write %x = %x\n", reg, value); + + ret = i2c_master_send(i2c, data, 3); + if (ret == 3) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r) +{ + struct i2c_msg xfer[2]; + u8 reg[2]; + u8 data; + int ret; + + /* Write register */ + reg[0] = (r >> 8) & 0xff; + reg[1] = r & 0xff; + xfer[0].addr = i2c->addr; + xfer[0].flags = 0; + xfer[0].len = sizeof(reg); + xfer[0].buf = ®[0]; + + /* Read data */ + xfer[1].addr = i2c->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 1; + xfer[1].buf = &data; + + ret = i2c_transfer(i2c->adapter, xfer, 2); + if (ret != 2) { + dev_err(&i2c->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + dev_vdbg(&i2c->dev, "read %x from %x\n", data, r); + + return data; +} + +static void wm2000_reset(struct wm2000_priv *wm2000) +{ + struct i2c_client *i2c = wm2000->i2c; + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_ENG_CLR); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_CLR); + wm2000_write(i2c, WM2000_REG_ID1, 0); + + wm2000->anc_mode = ANC_OFF; +} + +static int wm2000_poll_bit(struct i2c_client *i2c, + unsigned int reg, u8 mask, int timeout) +{ + int val; + + val = wm2000_read(i2c, reg); + + while (!(val & mask) && --timeout) { + msleep(1); + val = wm2000_read(i2c, reg); + } + + if (timeout == 0) + return 0; + else + return 1; +} + +static int wm2000_power_up(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + int ret, timeout; + + BUG_ON(wm2000->anc_mode != ANC_OFF); + + dev_dbg(&i2c->dev, "Beginning power up\n"); + + if (!wm2000->mclk_div) { + dev_dbg(&i2c->dev, "Disabling MCLK divider\n"); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, + WM2000_MCLK_DIV2_ENA_CLR); + } else { + dev_dbg(&i2c->dev, "Enabling MCLK divider\n"); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, + WM2000_MCLK_DIV2_ENA_SET); + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_ENG_CLR); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_ENG_SET); + + /* Wait for ANC engine to become ready */ + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, + WM2000_ANC_ENG_IDLE, 1)) { + dev_err(&i2c->dev, "ANC engine failed to reset\n"); + return -ETIMEDOUT; + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_BOOT_COMPLETE, 1)) { + dev_err(&i2c->dev, "ANC engine failed to initialise\n"); + return -ETIMEDOUT; + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_SET); + + /* Open code download of the data since it is the only bulk + * write we do. */ + dev_dbg(&i2c->dev, "Downloading %d bytes\n", + wm2000->anc_download_size - 2); + + ret = i2c_master_send(i2c, wm2000->anc_download, + wm2000->anc_download_size); + if (ret < 0) { + dev_err(&i2c->dev, "i2c_transfer() failed: %d\n", ret); + return ret; + } + if (ret != wm2000->anc_download_size) { + dev_err(&i2c->dev, "i2c_transfer() failed, %d != %d\n", + ret, wm2000->anc_download_size); + return -EIO; + } + + dev_dbg(&i2c->dev, "Download complete\n"); + + if (analogue) { + timeout = 248; + wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4); + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_MOUSE_ENABLE | + WM2000_MODE_THERMAL_ENABLE); + } else { + timeout = 10; + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_MOUSE_ENABLE | + WM2000_MODE_THERMAL_ENABLE); + } + + ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); + if (wm2000->speech_clarity) + ret &= ~WM2000_SPEECH_CLARITY; + else + ret |= WM2000_SPEECH_CLARITY; + wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); + + wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); + wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02); + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_MOUSE_ACTIVE, timeout)) { + dev_err(&i2c->dev, "Timed out waiting for device after %dms\n", + timeout * 10); + return -ETIMEDOUT; + } + + dev_dbg(&i2c->dev, "ANC active\n"); + if (analogue) + dev_dbg(&i2c->dev, "Analogue active\n"); + wm2000->anc_mode = ANC_ACTIVE; + + return 0; +} + +static int wm2000_power_down(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + int timeout; + + if (analogue) { + timeout = 248; + wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_POWER_DOWN); + } else { + timeout = 10; + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_POWER_DOWN); + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_POWER_DOWN_COMPLETE, timeout)) { + dev_err(&i2c->dev, "Timeout waiting for ANC power down\n"); + return -ETIMEDOUT; + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, + WM2000_ANC_ENG_IDLE, 1)) { + dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n"); + return -ETIMEDOUT; + } + + dev_dbg(&i2c->dev, "powered off\n"); + wm2000->anc_mode = ANC_OFF; + + return 0; +} + +static int wm2000_enter_bypass(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + BUG_ON(wm2000->anc_mode != ANC_ACTIVE); + + if (analogue) { + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_BYPASS_ENTRY); + } else { + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_BYPASS_ENTRY); + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_ANC_DISABLED, 10)) { + dev_err(&i2c->dev, "Timeout waiting for ANC disable\n"); + return -ETIMEDOUT; + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, + WM2000_ANC_ENG_IDLE, 1)) { + dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n"); + return -ETIMEDOUT; + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL1, WM2000_SYS_STBY); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_CLR); + + wm2000->anc_mode = ANC_BYPASS; + dev_dbg(&i2c->dev, "bypass enabled\n"); + + return 0; +} + +static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + BUG_ON(wm2000->anc_mode != ANC_BYPASS); + + wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0); + + if (analogue) { + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_MOUSE_ENABLE | + WM2000_MODE_THERMAL_ENABLE); + } else { + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_MOUSE_ENABLE | + WM2000_MODE_THERMAL_ENABLE); + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_SET); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_MOUSE_ACTIVE, 10)) { + dev_err(&i2c->dev, "Timed out waiting for MOUSE\n"); + return -ETIMEDOUT; + } + + wm2000->anc_mode = ANC_ACTIVE; + dev_dbg(&i2c->dev, "MOUSE active\n"); + + return 0; +} + +static int wm2000_enter_standby(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + int timeout; + + BUG_ON(wm2000->anc_mode != ANC_ACTIVE); + + if (analogue) { + timeout = 248; + wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4); + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_STANDBY_ENTRY); + } else { + timeout = 10; + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_STANDBY_ENTRY); + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_ANC_DISABLED, timeout)) { + dev_err(&i2c->dev, + "Timed out waiting for ANC disable after 1ms\n"); + return -ETIMEDOUT; + } + + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, WM2000_ANC_ENG_IDLE, + 1)) { + dev_err(&i2c->dev, + "Timed out waiting for standby after %dms\n", + timeout * 10); + return -ETIMEDOUT; + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL1, WM2000_SYS_STBY); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_CLR); + + wm2000->anc_mode = ANC_STANDBY; + dev_dbg(&i2c->dev, "standby\n"); + if (analogue) + dev_dbg(&i2c->dev, "Analogue disabled\n"); + + return 0; +} + +static int wm2000_exit_standby(struct i2c_client *i2c, int analogue) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + int timeout; + + BUG_ON(wm2000->anc_mode != ANC_STANDBY); + + wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0); + + if (analogue) { + timeout = 248; + wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4); + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_ANA_SEQ_INCLUDE | + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_MOUSE_ENABLE); + } else { + timeout = 10; + + wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, + WM2000_MODE_THERMAL_ENABLE | + WM2000_MODE_MOUSE_ENABLE); + } + + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_RAM_SET); + wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); + + if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, + WM2000_STATUS_MOUSE_ACTIVE, timeout)) { + dev_err(&i2c->dev, "Timed out waiting for MOUSE after %dms\n", + timeout * 10); + return -ETIMEDOUT; + } + + wm2000->anc_mode = ANC_ACTIVE; + dev_dbg(&i2c->dev, "MOUSE active\n"); + if (analogue) + dev_dbg(&i2c->dev, "Analogue enabled\n"); + + return 0; +} + +typedef int (*wm2000_mode_fn)(struct i2c_client *i2c, int analogue); + +static struct { + enum wm2000_anc_mode source; + enum wm2000_anc_mode dest; + int analogue; + wm2000_mode_fn step[2]; +} anc_transitions[] = { + { + .source = ANC_OFF, + .dest = ANC_ACTIVE, + .analogue = 1, + .step = { + wm2000_power_up, + }, + }, + { + .source = ANC_OFF, + .dest = ANC_STANDBY, + .step = { + wm2000_power_up, + wm2000_enter_standby, + }, + }, + { + .source = ANC_OFF, + .dest = ANC_BYPASS, + .analogue = 1, + .step = { + wm2000_power_up, + wm2000_enter_bypass, + }, + }, + { + .source = ANC_ACTIVE, + .dest = ANC_BYPASS, + .analogue = 1, + .step = { + wm2000_enter_bypass, + }, + }, + { + .source = ANC_ACTIVE, + .dest = ANC_STANDBY, + .analogue = 1, + .step = { + wm2000_enter_standby, + }, + }, + { + .source = ANC_ACTIVE, + .dest = ANC_OFF, + .analogue = 1, + .step = { + wm2000_power_down, + }, + }, + { + .source = ANC_BYPASS, + .dest = ANC_ACTIVE, + .analogue = 1, + .step = { + wm2000_exit_bypass, + }, + }, + { + .source = ANC_BYPASS, + .dest = ANC_STANDBY, + .analogue = 1, + .step = { + wm2000_exit_bypass, + wm2000_enter_standby, + }, + }, + { + .source = ANC_BYPASS, + .dest = ANC_OFF, + .step = { + wm2000_exit_bypass, + wm2000_power_down, + }, + }, + { + .source = ANC_STANDBY, + .dest = ANC_ACTIVE, + .analogue = 1, + .step = { + wm2000_exit_standby, + }, + }, + { + .source = ANC_STANDBY, + .dest = ANC_BYPASS, + .analogue = 1, + .step = { + wm2000_exit_standby, + wm2000_enter_bypass, + }, + }, + { + .source = ANC_STANDBY, + .dest = ANC_OFF, + .step = { + wm2000_exit_standby, + wm2000_power_down, + }, + }, +}; + +static int wm2000_anc_transition(struct wm2000_priv *wm2000, + enum wm2000_anc_mode mode) +{ + struct i2c_client *i2c = wm2000->i2c; + int i, j; + int ret; + + if (wm2000->anc_mode == mode) + return 0; + + for (i = 0; i < ARRAY_SIZE(anc_transitions); i++) + if (anc_transitions[i].source == wm2000->anc_mode && + anc_transitions[i].dest == mode) + break; + if (i == ARRAY_SIZE(anc_transitions)) { + dev_err(&i2c->dev, "No transition for %d->%d\n", + wm2000->anc_mode, mode); + return -EINVAL; + } + + for (j = 0; j < ARRAY_SIZE(anc_transitions[j].step); j++) { + if (!anc_transitions[i].step[j]) + break; + ret = anc_transitions[i].step[j](i2c, + anc_transitions[i].analogue); + if (ret != 0) + return ret; + } + + return 0; +} + +static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) +{ + struct i2c_client *i2c = wm2000->i2c; + enum wm2000_anc_mode mode; + + if (wm2000->anc_eng_ena && wm2000->spk_ena) + if (wm2000->anc_active) + mode = ANC_ACTIVE; + else + mode = ANC_BYPASS; + else + mode = ANC_STANDBY; + + dev_dbg(&i2c->dev, "Set mode %d (enabled %d, mute %d, active %d)\n", + mode, wm2000->anc_eng_ena, !wm2000->spk_ena, + wm2000->anc_active); + + return wm2000_anc_transition(wm2000, mode); +} + +static int wm2000_anc_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + + ucontrol->value.enumerated.item[0] = wm2000->anc_active; + + return 0; +} + +static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + int anc_active = ucontrol->value.enumerated.item[0]; + + if (anc_active > 1) + return -EINVAL; + + wm2000->anc_active = anc_active; + + return wm2000_anc_set_mode(wm2000); +} + +static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + + ucontrol->value.enumerated.item[0] = wm2000->spk_ena; + + return 0; +} + +static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + int val = ucontrol->value.enumerated.item[0]; + + if (val > 1) + return -EINVAL; + + wm2000->spk_ena = val; + + return wm2000_anc_set_mode(wm2000); +} + +static const struct snd_kcontrol_new wm2000_controls[] = { + SOC_SINGLE_BOOL_EXT("WM2000 ANC Switch", 0, + wm2000_anc_mode_get, + wm2000_anc_mode_put), + SOC_SINGLE_BOOL_EXT("WM2000 Switch", 0, + wm2000_speaker_get, + wm2000_speaker_put), +}; + +static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&wm2000_i2c->dev); + + if (SND_SOC_DAPM_EVENT_ON(event)) + wm2000->anc_eng_ena = 1; + + if (SND_SOC_DAPM_EVENT_OFF(event)) + wm2000->anc_eng_ena = 0; + + return wm2000_anc_set_mode(wm2000); +} + +static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { +/* Externally visible pins */ +SND_SOC_DAPM_OUTPUT("WM2000 SPKN"), +SND_SOC_DAPM_OUTPUT("WM2000 SPKP"), + +SND_SOC_DAPM_INPUT("WM2000 LINN"), +SND_SOC_DAPM_INPUT("WM2000 LINP"), + +SND_SOC_DAPM_PGA_E("ANC Engine", SND_SOC_NOPM, 0, 0, NULL, 0, + wm2000_anc_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +}; + +/* Target, Path, Source */ +static const struct snd_soc_dapm_route audio_map[] = { + { "WM2000 SPKN", NULL, "ANC Engine" }, + { "WM2000 SPKP", NULL, "ANC Engine" }, + { "ANC Engine", NULL, "WM2000 LINN" }, + { "ANC Engine", NULL, "WM2000 LINP" }, +}; + +/* Called from the machine driver */ +int wm2000_add_controls(struct snd_soc_codec *codec) +{ + int ret; + + if (!wm2000_i2c) { + pr_err("WM2000 not yet probed\n"); + return -ENODEV; + } + + ret = snd_soc_dapm_new_controls(codec, wm2000_dapm_widgets, + ARRAY_SIZE(wm2000_dapm_widgets)); + if (ret < 0) + return ret; + + ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + if (ret < 0) + return ret; + + return snd_soc_add_controls(codec, wm2000_controls, + ARRAY_SIZE(wm2000_controls)); +} +EXPORT_SYMBOL_GPL(wm2000_add_controls); + +static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct wm2000_priv *wm2000; + struct wm2000_platform_data *pdata; + const char *filename; + const struct firmware *fw; + int reg, ret; + u16 id; + + if (wm2000_i2c) { + dev_err(&i2c->dev, "Another WM2000 is already registered\n"); + return -EINVAL; + } + + wm2000 = kzalloc(sizeof(struct wm2000_priv), GFP_KERNEL); + if (wm2000 == NULL) { + dev_err(&i2c->dev, "Unable to allocate private data\n"); + return -ENOMEM; + } + + /* Verify that this is a WM2000 */ + reg = wm2000_read(i2c, WM2000_REG_ID1); + id = reg << 8; + reg = wm2000_read(i2c, WM2000_REG_ID2); + id |= reg & 0xff; + + if (id != 0x2000) { + dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id); + ret = -ENODEV; + goto err; + } + + reg = wm2000_read(i2c, WM2000_REG_REVISON); + dev_info(&i2c->dev, "revision %c\n", reg + 'A'); + + filename = "wm2000_anc.bin"; + pdata = dev_get_platdata(&i2c->dev); + if (pdata) { + wm2000->mclk_div = pdata->mclkdiv2; + wm2000->speech_clarity = !pdata->speech_enh_disable; + + if (pdata->download_file) + filename = pdata->download_file; + } + + ret = request_firmware(&fw, filename, &i2c->dev); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret); + goto err; + } + + /* Pre-cook the concatenation of the register address onto the image */ + wm2000->anc_download_size = fw->size + 2; + wm2000->anc_download = kmalloc(wm2000->anc_download_size, GFP_KERNEL); + if (wm2000->anc_download == NULL) { + dev_err(&i2c->dev, "Out of memory\n"); + ret = -ENOMEM; + goto err_fw; + } + + wm2000->anc_download[0] = 0x80; + wm2000->anc_download[1] = 0x00; + memcpy(wm2000->anc_download + 2, fw->data, fw->size); + + release_firmware(fw); + + dev_set_drvdata(&i2c->dev, wm2000); + wm2000->anc_eng_ena = 1; + wm2000->i2c = i2c; + + wm2000_reset(wm2000); + + /* This will trigger a transition to standby mode by default */ + wm2000_anc_set_mode(wm2000); + + wm2000_i2c = i2c; + + return 0; + +err_fw: + release_firmware(fw); +err: + kfree(wm2000); + return ret; +} + +static __devexit int wm2000_i2c_remove(struct i2c_client *i2c) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + wm2000_anc_transition(wm2000, ANC_OFF); + + wm2000_i2c = NULL; + kfree(wm2000->anc_download); + kfree(wm2000); + + return 0; +} + +static void wm2000_i2c_shutdown(struct i2c_client *i2c) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + wm2000_anc_transition(wm2000, ANC_OFF); +} + +#ifdef CONFIG_PM +static int wm2000_i2c_suspend(struct i2c_client *i2c, pm_message_t mesg) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + return wm2000_anc_transition(wm2000, ANC_OFF); +} + +static int wm2000_i2c_resume(struct i2c_client *i2c) +{ + struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + + return wm2000_anc_set_mode(wm2000); +} +#else +#define wm2000_i2c_suspend NULL +#define wm2000_i2c_resume NULL +#endif + +static const struct i2c_device_id wm2000_i2c_id[] = { + { "wm2000", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm2000_i2c_id); + +static struct i2c_driver wm2000_i2c_driver = { + .driver = { + .name = "wm2000", + .owner = THIS_MODULE, + }, + .probe = wm2000_i2c_probe, + .remove = __devexit_p(wm2000_i2c_remove), + .suspend = wm2000_i2c_suspend, + .resume = wm2000_i2c_resume, + .shutdown = wm2000_i2c_shutdown, + .id_table = wm2000_i2c_id, +}; + +static int __init wm2000_init(void) +{ + return i2c_add_driver(&wm2000_i2c_driver); +} +module_init(wm2000_init); + +static void __exit wm2000_exit(void) +{ + i2c_del_driver(&wm2000_i2c_driver); +} +module_exit(wm2000_exit); + +MODULE_DESCRIPTION("ASoC WM2000 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm2000.h b/sound/soc/codecs/wm2000.h new file mode 100644 index 0000000..c18e261 --- /dev/null +++ b/sound/soc/codecs/wm2000.h @@ -0,0 +1,79 @@ +/* + * wm2000.h -- WM2000 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM2000_H +#define _WM2000_H + +struct wm2000_setup_data { + unsigned short i2c_address; + int mclk_div; /* Set to a non-zero value if MCLK_DIV_2 required */ +}; + +extern int wm2000_add_controls(struct snd_soc_codec *codec); + +extern struct snd_soc_dai wm2000_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm2000; + +#define WM2000_REG_SYS_START 0x8000 +#define WM2000_REG_SPEECH_CLARITY 0x8fef +#define WM2000_REG_SYS_WATCHDOG 0x8ff6 +#define WM2000_REG_ANA_VMID_PD_TIME 0x8ff7 +#define WM2000_REG_ANA_VMID_PU_TIME 0x8ff8 +#define WM2000_REG_CAT_FLTR_INDX 0x8ff9 +#define WM2000_REG_CAT_GAIN_0 0x8ffa +#define WM2000_REG_SYS_STATUS 0x8ffc +#define WM2000_REG_SYS_MODE_CNTRL 0x8ffd +#define WM2000_REG_SYS_START0 0x8ffe +#define WM2000_REG_SYS_START1 0x8fff +#define WM2000_REG_ID1 0xf000 +#define WM2000_REG_ID2 0xf001 +#define WM2000_REG_REVISON 0xf002 +#define WM2000_REG_SYS_CTL1 0xf003 +#define WM2000_REG_SYS_CTL2 0xf004 +#define WM2000_REG_ANC_STAT 0xf005 +#define WM2000_REG_IF_CTL 0xf006 + +/* SPEECH_CLARITY */ +#define WM2000_SPEECH_CLARITY 0x01 + +/* SYS_STATUS */ +#define WM2000_STATUS_MOUSE_ACTIVE 0x40 +#define WM2000_STATUS_CAT_FREQ_COMPLETE 0x20 +#define WM2000_STATUS_CAT_GAIN_COMPLETE 0x10 +#define WM2000_STATUS_THERMAL_SHUTDOWN_COMPLETE 0x08 +#define WM2000_STATUS_ANC_DISABLED 0x04 +#define WM2000_STATUS_POWER_DOWN_COMPLETE 0x02 +#define WM2000_STATUS_BOOT_COMPLETE 0x01 + +/* SYS_MODE_CNTRL */ +#define WM2000_MODE_ANA_SEQ_INCLUDE 0x80 +#define WM2000_MODE_MOUSE_ENABLE 0x40 +#define WM2000_MODE_CAT_FREQ_ENABLE 0x20 +#define WM2000_MODE_CAT_GAIN_ENABLE 0x10 +#define WM2000_MODE_BYPASS_ENTRY 0x08 +#define WM2000_MODE_STANDBY_ENTRY 0x04 +#define WM2000_MODE_THERMAL_ENABLE 0x02 +#define WM2000_MODE_POWER_DOWN 0x01 + +/* SYS_CTL1 */ +#define WM2000_SYS_STBY 0x01 + +/* SYS_CTL2 */ +#define WM2000_MCLK_DIV2_ENA_CLR 0x80 +#define WM2000_MCLK_DIV2_ENA_SET 0x40 +#define WM2000_ANC_ENG_CLR 0x20 +#define WM2000_ANC_ENG_SET 0x10 +#define WM2000_ANC_INT_N_CLR 0x08 +#define WM2000_ANC_INT_N_SET 0x04 +#define WM2000_RAM_CLR 0x02 +#define WM2000_RAM_SET 0x01 + +/* ANC_STAT */ +#define WM2000_ANC_ENG_IDLE 0x01 + +#endif -- cgit v0.10.2 From cfd3d8dcf7b4fc783db0806ac3936a7b44735bf7 Mon Sep 17 00:00:00 2001 From: Greg Alexander Date: Sat, 13 Feb 2010 02:02:25 -0500 Subject: ALSA: hda - Add support for Lenovo IdeaPad U150 Add patch for the Conexant 5066 HDA codec to support the Lenovo IdeaPad U150 Signed-off-by: Greg Alexander Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 8f06f20..0c7ebef 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -288,6 +288,7 @@ Conexant 5066 laptop Basic Laptop config (default) dell-laptop Dell laptops olpc-xo-1_5 OLPC XO 1.5 + ideapad Lenovo IdeaPad U150 STAC9200 ======== diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 745e359..194a28c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -113,7 +113,8 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; - unsigned int dell_vostro; + unsigned int dell_vostro:1; + unsigned int ideapad:1; unsigned int ext_mic_present; unsigned int recording; @@ -2167,6 +2168,34 @@ static void cxt5066_vostro_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_ideapad_automic(struct hda_codec *codec) +{ + unsigned int present; + + struct hda_verb ext_mic_present[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 2}, + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + + present = snd_hda_jack_detect(codec, 0x1b); + if (present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + /* mute internal speaker if HP is plugged */ static void cxt5066_hp_automute(struct hda_codec *codec) { @@ -2216,6 +2245,20 @@ static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res) } } +/* unsolicited event for jack sensing */ +static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_ideapad_automic(codec); + break; + } +} + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -2227,13 +2270,21 @@ static const struct hda_input_mux cxt5066_analog_mic_boost = { }, }; -static int cxt5066_set_mic_boost(struct hda_codec *codec) +static void cxt5066_set_mic_boost(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - return snd_hda_codec_write_cache(codec, 0x17, 0, + snd_hda_codec_write_cache(codec, 0x17, 0, AC_VERB_SET_AMP_GAIN_MUTE, AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | cxt5066_analog_mic_boost.items[spec->mic_boost].index); + if (spec->ideapad) { + /* adjust the internal mic as well...it is not through 0x17 */ + snd_hda_codec_write_cache(codec, 0x23, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_INPUT | + cxt5066_analog_mic_boost. + items[spec->mic_boost].index); + } } static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, @@ -2664,6 +2715,56 @@ static struct hda_verb cxt5066_init_verbs_vostro[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_ideapad[] = { + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ + + /* Speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* HP, Amp */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ + + /* Audio input selector */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, + {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ + + /* SPDIF route: PCM */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, + + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* internal microphone */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable int mic */ + + /* EAPD */ + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + static struct hda_verb cxt5066_init_verbs_portd_lo[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { } /* end */ @@ -2680,6 +2781,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_hp_automute(codec); if (spec->dell_vostro) cxt5066_vostro_automic(codec); + else if (spec->ideapad) + cxt5066_ideapad_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -2705,6 +2808,7 @@ enum { CXT5066_DELL_LAPTOP, /* Dell Laptop */ CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ + CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_MODELS }; @@ -2712,7 +2816,8 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", - [CXT5066_DELL_VOSTO] = "dell-vostro" + [CXT5066_DELL_VOSTO] = "dell-vostro", + [CXT5066_IDEAPAD] = "ideapad", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -2722,6 +2827,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} }; @@ -2813,6 +2919,22 @@ static int patch_cxt5066(struct hda_codec *codec) /* input source automatically selected */ spec->input_mux = NULL; break; + case CXT5066_IDEAPAD: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_ideapad_event; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->init_verbs[0] = cxt5066_init_verbs_ideapad; + spec->port_d_mode = 0; + spec->ideapad = 1; + spec->mic_boost = 2; /* default 20dB gain */ + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + + /* input source automatically selected */ + spec->input_mux = NULL; + break; } return 0; -- cgit v0.10.2 From 19b50063780953563e3c3a2867c39aad7b9e64cf Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:34 +0100 Subject: ALSA: Echoaudio - Add firmware cache #1 Changes the way the firmware is passed through functions. When CONFIG_PM is enabled the firmware cannot be released because the driver will need it again to resume the card. With this patch the firmware is passed as an index of the struct firmware card_fw[] in place of a pointer. That same index is then used to locate the firmware in the firmware cache. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index 2904330..a44135d 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP]; + chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index 6022873..d681da1 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP]; + chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index 57967e5..f007193 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -61,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP]; + chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get what type of external box is attached */ diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1305f7c..78fc263 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -36,11 +36,15 @@ MODULE_PARM_DESC(enable, "Enable " ECHOCARD_NAME " soundcard."); static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999}; static const DECLARE_TLV_DB_SCALE(db_scale_output_gain, -12800, 100, 1); + + static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip) + struct echoaudio *chip, const short fw_index) { int err; char name[30]; + const struct firmware *frm = &card_fw[fw_index]; + DE_ACT(("firmware requested: %s\n", frm->data)); snprintf(name, sizeof(name), "ea/%s", frm->data); if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) @@ -48,6 +52,8 @@ static int get_firmware(const struct firmware **fw_entry, return err; } + + static void free_firmware(const struct firmware *fw_entry) { release_firmware(fw_entry); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index f9490ae..be76ef3 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -442,8 +442,8 @@ struct echoaudio { u16 device_id, subdevice_id; u16 *dsp_code; /* Current DSP code loaded, * NULL if nothing loaded */ - const struct firmware *dsp_code_to_load;/* DSP code to load */ - const struct firmware *asic_code; /* Current ASIC code */ + short dsp_code_to_load; /* DSP code to load */ + short asic_code; /* Current ASIC code */ u32 comm_page_phys; /* Physical address of the * memory seen by DSP */ volatile u32 __iomem *dsp_registers; /* DSP's register base */ @@ -464,7 +464,7 @@ static int load_firmware(struct echoaudio *chip); static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip); + struct echoaudio *chip, const short fw_index); static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index e32a748..658db44 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -227,12 +227,11 @@ static int load_asic(struct echoaudio *chip) /* Give the DSP a few milliseconds to settle down */ mdelay(2); - err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, - &card_fw[FW_3G_ASIC]); + err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, FW_3G_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_3G_ASIC]; + chip->asic_code = FW_3G_ASIC; /* Now give the new ASIC some time to set up */ msleep(1000); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 4df51ef..031ef7e 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -175,15 +175,15 @@ static inline int check_asic_status(struct echoaudio *chip) #ifdef ECHOCARD_HAS_ASIC /* Load ASIC code - done after the DSP is loaded */ -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic) +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) { const struct firmware *fw; int err; u32 i, size; u8 *code; - if ((err = get_firmware(&fw, asic, chip)) < 0) { + err = get_firmware(&fw, chip, asic); + if (err < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return err; } @@ -245,7 +245,8 @@ static int install_resident_loader(struct echoaudio *chip) return 0; } - if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) { + i = get_firmware(&fw, chip, FW_361_LOADER); + if (i < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return i; } @@ -485,7 +486,8 @@ static int load_firmware(struct echoaudio *chip) chip->dsp_code = NULL; } - if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0) + err = get_firmware(&fw, chip, chip->dsp_code_to_load); + if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); free_firmware(fw); diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index 3f1e747..c5de88b 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -49,7 +49,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP]; + chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 2fef37a..093dd7b 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -33,8 +33,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,13 +63,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { - chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP]; + chip->dsp_code_to_load = FW_GINA24_361_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; } else { - chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP]; + chip->dsp_code_to_load = FW_GINA24_301_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | @@ -125,7 +124,7 @@ static int load_asic(struct echoaudio *chip) { u32 control_reg; int err; - const struct firmware *fw; + short asic; if (chip->asic_loaded) return 1; @@ -135,14 +134,15 @@ static int load_asic(struct echoaudio *chip) /* Pick the correct ASIC for '301 or '361 Gina24 */ if (chip->device_id == DEVICE_ID_56361) - fw = &card_fw[FW_GINA24_361_ASIC]; + asic = FW_GINA24_361_ASIC; else - fw = &card_fw[FW_GINA24_301_ASIC]; + asic = FW_GINA24_301_ASIC; - if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0) + err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, asic); + if (err < 0) return err; - chip->asic_code = fw; + chip->asic_code = asic; /* Now give the new ASIC a little time to set up */ mdelay(10); diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 0b2cd9c..8799d2e 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 0839291..cb1c92c 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index f591fc2..91dbfeb 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 0604c8a..134e783 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index f357521..766cf50 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 83750e9..07f3245 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -31,8 +31,7 @@ static int read_dsp(struct echoaudio *chip, u32 *data); static int set_professional_spdif(struct echoaudio *chip, char prof); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); static int update_flags(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP]; + chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER; @@ -144,7 +143,7 @@ static int load_asic(struct echoaudio *chip) return 0; err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC, - &card_fw[FW_LAYLA20_ASIC]); + FW_LAYLA20_ASIC); if (err < 0) return err; diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index d61b5cb..12dc00a 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -32,8 +32,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP]; + chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -123,18 +122,18 @@ static int load_asic(struct echoaudio *chip) /* Load the ASIC for the PCI card */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC, - &card_fw[FW_LAYLA24_1_ASIC]); + FW_LAYLA24_1_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC]; + chip->asic_code = FW_LAYLA24_2S_ASIC; /* Now give the new ASIC a little time to set up */ mdelay(10); /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, - &card_fw[FW_LAYLA24_2S_ASIC]); + FW_LAYLA24_2S_ASIC); if (err < 0) return FALSE; @@ -299,7 +298,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) /* Depending on what digital mode you want, Layla24 needs different ASICs loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ -static int switch_asic(struct echoaudio *chip, const struct firmware *asic) +static int switch_asic(struct echoaudio *chip, short asic) { s8 *monitors; @@ -335,7 +334,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) { u32 control_reg; int err, incompatible_clock; - const struct firmware *asic; + short asic; /* Set clock to "internal" if it's not compatible with the new mode */ incompatible_clock = FALSE; @@ -344,12 +343,12 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2S_ASIC]; + asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2A_ASIC]; + asic = FW_LAYLA24_2A_ASIC; break; default: DE_ACT(("Digital mode not supported: %d\n", mode)); diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index 5514051..d0302f2 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -53,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_MIA_DSP]; + chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index eaa619b..b28b8e4 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -33,8 +33,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,9 +63,9 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) /* Mona comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) - chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP]; + chip->dsp_code_to_load = FW_MONA_361_DSP; else - chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP]; + chip->dsp_code_to_load = FW_MONA_301_DSP; chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; chip->professional_spdif = FALSE; @@ -120,7 +119,7 @@ static int load_asic(struct echoaudio *chip) { u32 control_reg; int err; - const struct firmware *asic; + short asic; if (chip->asic_loaded) return 0; @@ -128,9 +127,9 @@ static int load_asic(struct echoaudio *chip) mdelay(10); if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic); if (err < 0) @@ -141,7 +140,7 @@ static int load_asic(struct echoaudio *chip) /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC, - &card_fw[FW_MONA_2_ASIC]); + FW_MONA_2_ASIC); if (err < 0) return err; @@ -165,22 +164,22 @@ loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ static int switch_asic(struct echoaudio *chip, char double_speed) { - const struct firmware *asic; int err; + short asic; /* Check the clock detect bits to see if this is a single-speed clock or a double-speed clock; load a new ASIC if necessary. */ if (chip->device_id == DEVICE_ID_56361) { if (double_speed) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; } else { if (double_speed) - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } if (asic != chip->asic_code) { @@ -200,7 +199,7 @@ static int switch_asic(struct echoaudio *chip, char double_speed) static int set_sample_rate(struct echoaudio *chip, u32 rate) { u32 control_reg, clock; - const struct firmware *asic; + short asic; char force_write; /* Only set the clock for internal mode. */ @@ -218,14 +217,14 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EINVAL; if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; } else { if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } force_write = 0; @@ -410,8 +409,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) case DIGITAL_MODE_ADAT: /* If the current ASIC is the 96KHz ASIC, switch the ASIC and set to 48 KHz */ - if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] || - chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) { + if (chip->asic_code == FW_MONA_361_1_ASIC96 || + chip->asic_code == FW_MONA_301_1_ASIC96) { set_sample_rate(chip, 48000); } control_reg |= GML_ADAT_MODE; -- cgit v0.10.2 From 4f8ada444cc7a7ea70cdc81f098b34c5f1f2df41 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:51 +0100 Subject: ALSA: Echoaudio - Add firmware cache #2 This patch implements a simple cache for the firmware files when CONFIG_PM is defined. This patch changes get_firmware(), free_firmware() and adds free_firmware_cache(). The first two functions implement a very simple cache and the latter is used to actually release all the stored firmwares when the module is unloaded. When CONFIG_PM is not enabled those functions act as before, that is free_firmware() releases the firmware immediately and free_firmware_cache() does nothing. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 78fc263..79dde95 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -43,12 +43,24 @@ static int get_firmware(const struct firmware **fw_entry, { int err; char name[30]; - const struct firmware *frm = &card_fw[fw_index]; - DE_ACT(("firmware requested: %s\n", frm->data)); - snprintf(name, sizeof(name), "ea/%s", frm->data); - if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) +#ifdef CONFIG_PM + if (chip->fw_cache[fw_index]) { + DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data)); + *fw_entry = chip->fw_cache[fw_index]; + return 0; + } +#endif + + DE_ACT(("firmware requested: %s\n", card_fw[fw_index].data)); + snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); + err = request_firmware(fw_entry, name, pci_device(chip)); + if (err < 0) snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err); +#ifdef CONFIG_PM + else + chip->fw_cache[fw_index] = *fw_entry; +#endif return err; } @@ -56,8 +68,29 @@ static int get_firmware(const struct firmware **fw_entry, static void free_firmware(const struct firmware *fw_entry) { +#ifdef CONFIG_PM + DE_ACT(("firmware not released (kept in cache)\n")); +#else release_firmware(fw_entry); DE_ACT(("firmware released\n")); +#endif +} + + + +static void free_firmware_cache(struct echoaudio *chip) +{ +#ifdef CONFIG_PM + int i; + + for (i = 0; i < 8 ; i++) + if (chip->fw_cache[i]) { + release_firmware(chip->fw_cache[i]); + DE_ACT(("release_firmware(%d)\n", i)); + } + + DE_ACT(("firmware_cache released\n")); +#endif } @@ -1880,6 +1913,7 @@ static int snd_echo_free(struct echoaudio *chip) pci_disable_device(chip->pci); /* release chip data */ + free_firmware_cache(chip); kfree(chip); DE_INIT(("Chip freed.\n")); return 0; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index be76ef3..a84c0d1 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -449,6 +449,9 @@ struct echoaudio { volatile u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ +#ifdef CONFIG_PM + const struct firmware *fw_cache[8]; /* Cached firmwares */ +#endif #ifdef ECHOCARD_HAS_MIDI u16 mtc_state; /* State for MIDI input parsing state machine */ -- cgit v0.10.2 From ad3499f4668f684ef6e5d0222ae14d5e4ade1fdd Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:59 +0100 Subject: ALSA: Echoaudio - Add suspend support #1 Move the controls init code outside the init_hw() function because is must not be called during resume. This patch moves the code that initializes the card's controls with default valued from the init_hw() function into a separated set_mixer_defaults() function (one for each of the 16 supported cards). This change is necessary because during resume we must resurrect the hardware without losing the previous settings. set_mixer_defaults() must be called only once when the module is loaded. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index a44135d..20c7cbc 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -57,15 +57,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + /* The Darla20 has no external clock sources */ static u32 detect_input_clocks(const struct echoaudio *chip) { diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index d681da1..6da6663 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -56,15 +56,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index f007193..3cdc2ee 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -97,20 +97,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_phantom_power(chip, 0); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); DE_INIT(("init_hw done\n")); return err; @@ -118,6 +104,18 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->non_audio_spdif = FALSE; + chip->bad_board = FALSE; + chip->phantom_power = FALSE; + return init_line_levels(chip); +} + + + static int set_phantom_power(struct echoaudio *chip, char on) { u32 control_reg = le32_to_cpu(chip->comm_page->control_register); diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index c5de88b..d1615a0 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -62,17 +62,20 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 093dd7b..98f7cfa 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -57,9 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | ECHO_CLOCK_BIT_ADAT; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { @@ -81,19 +78,22 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 8799d2e..5e85f14 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c index 9ab625e..2e4ab3e 100644 --- a/sound/pci/echoaudio/indigo_express_dsp.c +++ b/sound/pci/echoaudio/indigo_express_dsp.c @@ -61,6 +61,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) control_reg |= clock; if (control_reg != old_control_reg) { + DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); chip->comm_page->control_register = cpu_to_le32(control_reg); chip->sample_rate = rate; clear_handshake(chip); diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index cb1c92c..68f3c8c 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 91dbfeb..bb9632c7 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 134e783..beb9a5b 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 766cf50..394c6e76 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 07f3245..53ce946 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -64,17 +64,20 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index 12dc00a..8c04164 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -61,9 +61,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; if ((err = load_firmware(chip)) < 0) return err; @@ -72,17 +69,22 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index d0302f2..6ebfa6e 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -66,15 +66,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip))) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index b28b8e4..6e6a7eb 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -67,28 +67,26 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) else chip->dsp_code_to_load = FW_MONA_301_DSP; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - if ((err = load_firmware(chip)) < 0) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; -- cgit v0.10.2 From 47b5d028fdce8f809bf22852ac900338fb90e8aa Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:16:10 +0100 Subject: ALSA: Echoaudio - Add suspend support #2 This patch adds rearranges parts of the initialization code and adds suspend and resume callbacks. This patch adds suspend and resume callbacks. It also rearranges parts of the initialization code so it can be used in both the first initialization (when the module is loaded we also have to load default settings) and the resume callback (where we have to restore the previous settings). Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 79dde95..2783ce6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -753,6 +753,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + DE_ACT(("pcm_trigger resume\n")); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: DE_ACT(("pcm_trigger start\n")); @@ -776,6 +778,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = start_transport(chip, channelmask, chip->pipe_cyclic_mask); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + DE_ACT(("pcm_trigger suspend\n")); case SNDRV_PCM_TRIGGER_STOP: DE_ACT(("pcm_trigger stop\n")); for (i = 0; i < DSP_MAXPIPES; i++) { @@ -1951,18 +1955,27 @@ static __devinit int snd_echo_create(struct snd_card *card, return err; pci_set_master(pci); - /* allocate a chip-specific data */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (!chip) { - pci_disable_device(pci); - return -ENOMEM; + /* Allocate chip if needed */ + if (!*rchip) { + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + pci_disable_device(pci); + return -ENOMEM; + } + DE_INIT(("chip=%p\n", chip)); + spin_lock_init(&chip->lock); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + atomic_set(&chip->opencount, 0); + mutex_init(&chip->mode_mutex); + chip->can_set_rate = 1; + } else { + /* If this was called from the resume function, chip is + * already allocated and it contains current card settings. + */ + chip = *rchip; } - DE_INIT(("chip=%p\n", chip)); - - spin_lock_init(&chip->lock); - chip->card = card; - chip->pci = pci; - chip->irq = -1; /* PCI resource allocation */ chip->dsp_registers_phys = pci_resource_start(pci, 0); @@ -2002,7 +2015,9 @@ static __devinit int snd_echo_create(struct snd_card *card, chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); - if (err) { + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { DE_INIT(("init_hw err=%d\n", err)); snd_echo_free(chip); return err; @@ -2013,9 +2028,6 @@ static __devinit int snd_echo_create(struct snd_card *card, snd_echo_free(chip); return err; } - atomic_set(&chip->opencount, 0); - mutex_init(&chip->mode_mutex); - chip->can_set_rate = 1; *rchip = chip; /* Init done ! */ return 0; @@ -2048,6 +2060,7 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, snd_card_set_dev(card, &pci->dev); + chip = NULL; /* Tells snd_echo_create to allocate chip */ if ((err = snd_echo_create(card, pci, &chip)) < 0) { snd_card_free(card); return err; @@ -2187,6 +2200,112 @@ ctl_error: +#if defined(CONFIG_PM) + +static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + + DE_INIT(("suspend start\n")); + snd_pcm_suspend_all(chip->analog_pcm); + snd_pcm_suspend_all(chip->digital_pcm); + +#ifdef ECHOCARD_HAS_MIDI + /* This call can sleep */ + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 0); +#endif + spin_lock_irq(&chip->lock); + if (wait_handshake(chip)) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + clear_handshake(chip); + if (send_vector(chip, DSP_VC_GO_COMATOSE) < 0) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + spin_unlock_irq(&chip->lock); + + chip->dsp_code = NULL; + free_irq(chip->irq, chip); + chip->irq = -1; + pci_save_state(pci); + pci_disable_device(pci); + + DE_INIT(("suspend done\n")); + return 0; +} + + + +static int snd_echo_resume(struct pci_dev *pci) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); + commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + + err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); + snd_echo_free(chip); + return err; + } + DE_INIT(("resume init OK\n")); + + /* Temporarily set chip->pipe_alloc_mask=0 otherwise + * restore_dsp_settings() fails. + */ + pipe_alloc_mask = chip->pipe_alloc_mask; + chip->pipe_alloc_mask = 0; + err = restore_dsp_rettings(chip); + chip->pipe_alloc_mask = pipe_alloc_mask; + if (err < 0) { + kfree(commpage_bak); + return err; + } + DE_INIT(("resume restore OK\n")); + + memcpy(&commpage->audio_format, &commpage_bak->audio_format, + sizeof(commpage->audio_format)); + memcpy(&commpage->sglist_addr, &commpage_bak->sglist_addr, + sizeof(commpage->sglist_addr)); + memcpy(&commpage->midi_output, &commpage_bak->midi_output, + sizeof(commpage->midi_output)); + kfree(commpage_bak); + + if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, + ECHOCARD_NAME, chip)) { + snd_echo_free(chip); + snd_printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + DE_INIT(("resume irq=%d\n", chip->irq)); + +#ifdef ECHOCARD_HAS_MIDI + if (chip->midi_input_enabled) + enable_midi_input(chip, TRUE); + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 1); +#endif + + DE_INIT(("resume done\n")); + return 0; +} + +#endif /* CONFIG_PM */ + + + static void __devexit snd_echo_remove(struct pci_dev *pci) { struct echoaudio *chip; @@ -2209,6 +2328,10 @@ static struct pci_driver driver = { .id_table = snd_echo_ids, .probe = snd_echo_probe, .remove = __devexit_p(snd_echo_remove), +#ifdef CONFIG_PM + .suspend = snd_echo_suspend, + .resume = snd_echo_resume, +#endif /* CONFIG_PM */ }; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index a84c0d1..1df974d 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -472,6 +472,8 @@ static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); +static void snd_echo_midi_output_trigger( + struct snd_rawmidi_substream *substream, int up); static int midi_service_irq(struct echoaudio *chip); static int __devinit snd_echo_midi_create(struct snd_card *card, struct echoaudio *chip); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 031ef7e..64417a7 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -497,9 +497,6 @@ static int load_firmware(struct echoaudio *chip) if ((box_type = load_asic(chip)) < 0) return box_type; /* error */ - if ((err = restore_dsp_rettings(chip)) < 0) - return err; - return box_type; } @@ -659,51 +656,106 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { - int err; + int i, o, err; DE_INIT(("restore_dsp_settings\n")); if ((err = check_asic_status(chip)) < 0) return err; - /* @ Gina20/Darla20 only. Should be harmless for other cards. */ + /* Gina20/Darla20 only. Should be harmless for other cards. */ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF; chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF; chip->comm_page->handshake = 0xffffffff; - if ((err = set_sample_rate(chip, chip->sample_rate)) < 0) + /* Restore output busses */ + for (i = 0; i < num_busses_out(chip); i++) { + err = set_output_gain(chip, i, chip->output_gain[i]); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_VMIXER + for (i = 0; i < num_pipes_out(chip); i++) + for (o = 0; o < num_busses_out(chip); o++) { + err = set_vmixer_gain(chip, o, i, + chip->vmixer_gain[o][i]); + if (err < 0) + return err; + } + if (update_vmixer_level(chip) < 0) + return -EIO; +#endif /* ECHOCARD_HAS_VMIXER */ + +#ifdef ECHOCARD_HAS_MONITOR + for (o = 0; o < num_busses_out(chip); o++) + for (i = 0; i < num_busses_in(chip); i++) { + err = set_monitor_gain(chip, o, i, + chip->monitor_gain[o][i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_MONITOR */ + +#ifdef ECHOCARD_HAS_INPUT_GAIN + for (i = 0; i < num_busses_in(chip); i++) { + err = set_input_gain(chip, i, chip->input_gain[i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_INPUT_GAIN */ + + err = update_output_line_level(chip); + if (err < 0) return err; - if (chip->meters_enabled) - if (send_vector(chip, DSP_VC_METERS_ON) < 0) - return -EIO; + err = update_input_line_level(chip); + if (err < 0) + return err; -#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK - if (set_input_clock(chip, chip->input_clock) < 0) + err = set_sample_rate(chip, chip->sample_rate); + if (err < 0) + return err; + + if (chip->meters_enabled) { + err = send_vector(chip, DSP_VC_METERS_ON); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH + if (set_digital_mode(chip, chip->digital_mode) < 0) return -EIO; #endif -#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH - if (set_output_clock(chip, chip->output_clock) < 0) +#ifdef ECHOCARD_HAS_DIGITAL_IO + if (set_professional_spdif(chip, chip->professional_spdif) < 0) return -EIO; #endif - if (update_output_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_PHANTOM_POWER + if (set_phantom_power(chip, chip->phantom_power) < 0) return -EIO; +#endif - if (update_input_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK + /* set_input_clock() also restores automute setting */ + if (set_input_clock(chip, chip->input_clock) < 0) return -EIO; +#endif -#ifdef ECHOCARD_HAS_VMIXER - if (update_vmixer_level(chip) < 0) +#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH + if (set_output_clock(chip, chip->output_clock) < 0) return -EIO; #endif if (wait_handshake(chip) < 0) return -EIO; clear_handshake(chip); + if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) + return -EIO; DE_INIT(("restore_dsp_rettings done\n")); - return send_vector(chip, DSP_VC_UPDATE_FLAGS); + return 0; } @@ -920,9 +972,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) chip->card_name = ECHOCARD_NAME; chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->digital_mode = DIGITAL_MODE_NONE; - chip->input_clock = ECHO_CLOCK_INTERNAL; - chip->output_clock = ECHO_CLOCK_WORD; chip->asic_loaded = FALSE; memset(chip->comm_page, 0, sizeof(struct comm_page)); @@ -933,7 +982,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) chip->comm_page->midi_out_free_count = cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); chip->comm_page->sample_rate = cpu_to_le32(44100); - chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE); @@ -944,50 +992,21 @@ static int init_dsp_comm_page(struct echoaudio *chip) -/* This function initializes the several volume controls for busses and pipes. -This MUST be called after the DSP is up and running ! */ +/* This function initializes the chip structure with default values, ie. all + * muted and internal clock source. Then it copies the settings to the DSP. + * This MUST be called after the DSP is up and running ! + */ static int init_line_levels(struct echoaudio *chip) { - int st, i, o; - DE_INIT(("init_line_levels\n")); - - /* Mute output busses */ - for (i = 0; i < num_busses_out(chip); i++) - if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; - -#ifdef ECHOCARD_HAS_VMIXER - /* Mute the Vmixer */ - for (i = 0; i < num_pipes_out(chip); i++) - for (o = 0; o < num_busses_out(chip); o++) - if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_vmixer_level(chip))) - return st; -#endif /* ECHOCARD_HAS_VMIXER */ - -#ifdef ECHOCARD_HAS_MONITOR - /* Mute the monitor mixer */ - for (o = 0; o < num_busses_out(chip); o++) - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_MONITOR */ - -#ifdef ECHOCARD_HAS_INPUT_GAIN - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_input_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_INPUT_GAIN */ - - return 0; + memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); + memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); + memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); + memset(chip->vmixer_gain, ECHOGAIN_MUTED, sizeof(chip->vmixer_gain)); + chip->input_clock = ECHO_CLOCK_INTERNAL; + chip->output_clock = ECHO_CLOCK_WORD; + chip->sample_rate = 44100; + return restore_dsp_rettings(chip); } -- cgit v0.10.2 From f167e1d073278fe231bbdd5d6c24fb9d091aa544 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 15 Feb 2010 08:55:28 +0100 Subject: ALSA: usb-audio: reduce MIDI packet size to work around broken firmware Extend the list of devices whose firmware does not expect more than one USB MIDI packet in one USB packet. bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752 Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6e89b83..aae50df 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1162,10 +1162,22 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, pipe = usb_sndintpipe(umidi->dev, ep_info->out_ep); else pipe = usb_sndbulkpipe(umidi->dev, ep_info->out_ep); - if (umidi->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ - ep->max_transfer = 4; - else + switch (umidi->usb_id) { + default: ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1); + break; + /* + * Various chips declare a packet size larger than 4 bytes, but + * do not actually work with larger packets: + */ + case USB_ID(0x0a92, 0x1020): /* ESI M4U */ + case USB_ID(0x1430, 0x474b): /* RedOctane GH MIDI INTERFACE */ + case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */ + case USB_ID(0x15ca, 0x1806): /* Textech USB Midi Cable */ + case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ + ep->max_transfer = 4; + break; + } for (i = 0; i < OUTPUT_URBS; ++i) { buffer = usb_buffer_alloc(umidi->dev, ep->max_transfer, GFP_KERNEL, -- cgit v0.10.2 From d39e82db73eb876c60d00f00219d767b3be30307 Mon Sep 17 00:00:00 2001 From: Sebastien Alaiwan Date: Tue, 16 Feb 2010 08:55:08 +0100 Subject: ALSA: USB MIDI support for Access Music VirusTI Here's a patch that adds MIDI support through USB for one of the Access Music synths, the VirusTI. The synth uses standard USBMIDI protocol on its USB interface 3, although it does signal "vendor specific" class. A magic string has to be sent on interface 3 to enable the sending of MIDI from the synth (this string was found by sniffing usb communication of the Windows driver). This is all my patch does, and it works on my computer. Please note that the synth can also do standard usb audio I/O on its interfaces 2&3, which already works with the current snd-usb-audio driver, except for the audio input from the synth. I'm going to work on it when I have some time. Signed-off-by: Sebastien Alaiwan Signed-off-by: Clemens Ladisch (cosmetics, list terminator) Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4963def..d01ec18 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3327,6 +3327,32 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) } /* + * This call will put the synth in "USB send" mode, i.e it will send MIDI + * messages through USB (this is disabled at startup). The synth will + * acknowledge by sending a sysex on endpoint 0x85 and by displaying a USB + * sign on its LCD. Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) +{ + int err, actual_length; + + /* "midi send" enable */ + static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; + + void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); + if (!buf) + return -ENOMEM; + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, + ARRAY_SIZE(seq), &actual_length, 1000); + kfree(buf); + if (err < 0) + return err; + + return 0; +} + +/* * Setup quirks */ #define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */ @@ -3624,6 +3650,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* Access Music VirusTI Desktop */ + if (id == USB_ID(0x133e, 0x0815)) { + if (snd_usb_accessmusic_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6e89b83..8f5bc1e 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1407,6 +1407,12 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Access Music Virus TI */ + EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), + PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER), }; static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index a892bda..406b74b 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2073,6 +2073,33 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Access Music devices */ +{ + /* VirusTI Desktop */ + USB_DEVICE_VENDOR_SPEC(0x133e, 0x0815), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + /* */ { /* aka. Serato Scratch Live DJ Box */ -- cgit v0.10.2 From ebfdeea3df2b8c265975b6acc47996a0b7c507e8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:17:09 +0100 Subject: ALSA: usbmixer - introduce /proc/asound/card#/usbmixer file The usbmixer proc file contains mapping between ALSA control API and USB mixer control units. The purpose of this file is for debugging and a problem diagnostics. Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index dd0c1d7..170bfd4 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -69,13 +69,16 @@ static const struct rc_config { { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; +#define MAX_ID_ELEMS 256 + struct usb_mixer_interface { struct snd_usb_audio *chip; unsigned int ctrlif; struct list_head list; unsigned int ignore_ctl_error; struct urb *urb; - struct usb_mixer_elem_info **id_elems; /* array[256], indexed by unit id */ + /* array[MAX_ID_ELEMS], indexed by unit id */ + struct usb_mixer_elem_info **id_elems; /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; @@ -1825,6 +1828,45 @@ static void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, info->elem_id); } +static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, + int unitid, + struct usb_mixer_elem_info *cval) +{ + static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", + "S8", "U8", "S16", "U16"}; + snd_iprintf(buffer, " Unit: %i\n", unitid); + if (cval->elem_id) + snd_iprintf(buffer, " Control: name=\"%s\", index=%i\n", + cval->elem_id->name, cval->elem_id->index); + snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " + "channels=%i, type=\"%s\"\n", cval->id, + cval->control, cval->cmask, cval->channels, + val_types[cval->val_type]); + snd_iprintf(buffer, " Volume: min=%i, max=%i, dBmin=%i, dBmax=%i\n", + cval->min, cval->max, cval->dBmin, cval->dBmax); +} + +static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_usb_audio *chip = entry->private_data; + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + snd_iprintf(buffer, + "USB Mixer: ctrlif=%i, ctlerr=%i\n", + mixer->ctrlif, mixer->ignore_ctl_error); + snd_iprintf(buffer, "Card: %s\n", chip->card->longname); + for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { + for (cval = mixer->id_elems[unitid]; cval; + cval = cval->next_id_elem) + snd_usb_mixer_dump_cval(buffer, unitid, cval); + } + } +} + static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, int unitid) { @@ -2187,20 +2229,21 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) } void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, - unsigned char samplerate_id) + unsigned char samplerate_id) { - struct usb_mixer_interface *mixer; - struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid = 12; /* SamleRate ExtensionUnit ID */ - list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer->id_elems[unitid]; - if (cval) { - set_cur_ctl_value(cval, cval->control << 8, samplerate_id); + list_for_each_entry(mixer, &chip->mixer_list, list) { + cval = mixer->id_elems[unitid]; + if (cval) { + set_cur_ctl_value(cval, cval->control << 8, + samplerate_id); snd_usb_mixer_notify_id(mixer, unitid); - } - break; - } + } + break; + } } int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, @@ -2210,6 +2253,7 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, .dev_free = snd_usb_mixer_dev_free }; struct usb_mixer_interface *mixer; + struct snd_info_entry *entry; int err; strcpy(chip->card->mixername, "USB Mixer"); @@ -2236,8 +2280,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { - struct snd_info_entry *entry; - if ((err = snd_audigy2nx_controls_create(mixer)) < 0) goto _error; if (!snd_card_proc_new(chip->card, "audigy2nx", &entry)) @@ -2255,6 +2297,11 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, err = snd_device_new(chip->card, SNDRV_DEV_LOWLEVEL, mixer, &dev_ops); if (err < 0) goto _error; + + if (list_empty(&chip->mixer_list) && + !snd_card_proc_new(chip->card, "usbmixer", &entry)) + snd_info_set_text_ops(entry, chip, snd_usb_mixer_proc_read); + list_add(&mixer->list, &chip->mixer_list); return 0; -- cgit v0.10.2 From 3be522a9514f58e0596db34898a514df206cadc5 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:55:43 +0100 Subject: ALSA: pcm core - fix fifo_size channels interval check Signed-off-by: Jaroslav Kysela Cc: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 25b0641..f7e1c9f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -315,10 +315,10 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!params->info) params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES; if (!params->fifo_size) { - if (snd_mask_min(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT]) == - snd_mask_max(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT]) && - snd_mask_min(¶ms->masks[SNDRV_PCM_HW_PARAM_CHANNELS]) == - snd_mask_max(¶ms->masks[SNDRV_PCM_HW_PARAM_CHANNELS])) { + m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + if (snd_mask_min(m) == snd_mask_max(m) && + snd_interval_min(i) == snd_interval_max(i)) { changed = substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_FIFO_SIZE, params); if (changed < 0) -- cgit v0.10.2 From 7affdc17d49b5d9e9c350d5d99ee34ab8655c7b4 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:52:27 +0100 Subject: ALSA: usbmixer - add usb_id value to usbmixer proc file Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 170bfd4..03f125d 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1856,8 +1856,9 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, list_for_each_entry(mixer, &chip->mixer_list, list) { snd_iprintf(buffer, - "USB Mixer: ctrlif=%i, ctlerr=%i\n", - mixer->ctrlif, mixer->ignore_ctl_error); + "USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n", + chip->usb_id, mixer->ctrlif, + mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { for (cval = mixer->id_elems[unitid]; cval; -- cgit v0.10.2 From 291186e049d7f8178ad31d43c38a53889f25d79e Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:55:18 +0100 Subject: ALSA: usbmixer - use MAX_ID_ELEMS where possible Signed-off-by: Jaroslav Kysela diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 03f125d..35b4830 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -108,7 +108,7 @@ struct mixer_build { struct usb_mixer_interface *mixer; unsigned char *buffer; unsigned int buflen; - DECLARE_BITMAP(unitbitmap, 256); + DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -2265,7 +2265,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, mixer->chip = chip; mixer->ctrlif = ctrlif; mixer->ignore_ctl_error = ignore_error; - mixer->id_elems = kcalloc(256, sizeof(*mixer->id_elems), GFP_KERNEL); + mixer->id_elems = kcalloc(MAX_ID_ELEMS, sizeof(*mixer->id_elems), + GFP_KERNEL); if (!mixer->id_elems) { kfree(mixer); return -ENOMEM; -- cgit v0.10.2 From 96dd362284ddcb546d2783035ae7eeda73692eda Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Feb 2010 11:05:44 +0000 Subject: ASoC: Make pmdown_time a per-card setting Make the pmdown_time a per-card setting rather than a global one, initialised before the card initialisation runs. This allows cards to override the default setting if it makes sense to do so (for example, due to an unavoidable pop). Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/soc.h b/include/sound/soc.h index e6a6d10..d9d88dd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -521,6 +521,8 @@ struct snd_soc_card { int (*set_bias_level)(struct snd_soc_card *, enum snd_soc_bias_level level); + int pmdown_time; + /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ca89c78..94b9cde 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -542,7 +542,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; schedule_delayed_work(&card->delayed_work, - msecs_to_jiffies(pmdown_time)); + msecs_to_jiffies(card->pmdown_time)); } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(codec, @@ -1039,6 +1039,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) dev_dbg(card->dev, "All components present, instantiating\n"); /* Found everything, bring it up */ + card->pmdown_time = pmdown_time; + if (card->probe) { ret = card->probe(pdev); if (ret < 0) -- cgit v0.10.2 From dbe21408b15f04da4f80fb89a27b7cb067d6103e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Feb 2010 11:37:24 +0000 Subject: ASoC: Make pmdown_time runtime configurable Provide a sysfs file allowing userspace to inspect and change the pmdown_time setting at runtime. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 94b9cde..c2008bc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -130,6 +130,29 @@ static ssize_t codec_reg_show(struct device *dev, static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +static ssize_t pmdown_time_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *socdev = dev_get_drvdata(dev); + struct snd_soc_card *card = socdev->card; + + return sprintf(buf, "%d\n", card->pmdown_time); +} + +static ssize_t pmdown_time_set(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_soc_device *socdev = dev_get_drvdata(dev); + struct snd_soc_card *card = socdev->card; + + strict_strtol(buf, 10, &card->pmdown_time); + + return count; +} + +static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); + #ifdef CONFIG_DEBUG_FS static int codec_reg_open_file(struct inode *inode, struct file *file) { @@ -1124,6 +1147,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); + ret = device_create_file(card->socdev->dev, &dev_attr_pmdown_time); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg); if (ret < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); -- cgit v0.10.2 From e5e878c1c393de917391477bc7627d729f7568fb Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 16 Feb 2010 13:23:15 +0200 Subject: ASoC: tlv320dac33: Clearing FIFOFLUSH flag before playback In repeated playback the FIFOFLUSH bit remained set, and never has been cleared. Clear it during the setup phase. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 1b35d0c..dab7fd5 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -734,7 +734,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); aictrl_a &= ~(DAC33_NCYCL_MASK | DAC33_WLEN_MASK); + /* Read FIFO control A, and clear FIFO flush bit */ fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + fifoctrl_a &= ~DAC33_FIFOFLUSH; + fifoctrl_a &= ~DAC33_WIDTH; switch (substream->runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: -- cgit v0.10.2 From 7833ae0edf50b0eb303e95b1bec5fbd63a1e2672 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 16 Feb 2010 13:23:16 +0200 Subject: ASoC: tlv320dac33: Correct the OSCSET calculation OSCSET calculation was not correct in case of 44.1KHz sampling rate. With small adjustment both 48 and 44.1 KHz calculation now gives the correct value. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index dab7fd5..f9f367d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -700,7 +700,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, } #define CALC_OSCSET(rate, refclk) ( \ - ((((rate * 10000) / refclk) * 4096) + 5000) / 10000) + ((((rate * 10000) / refclk) * 4096) + 7000) / 10000) #define CALC_RATIOSET(rate, refclk) ( \ ((((refclk * 100000) / rate) * 16384) + 50000) / 100000) -- cgit v0.10.2 From b721e68bdc5b39c51bf6a1469f8d3663fbe03243 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Wed, 17 Feb 2010 00:57:44 +0100 Subject: ALSA: Echoaudio, fix Guru Meditation #00000005.48454C50 This patch fixes a division by zero error in the irq handler. There is a small window between the hw_params() callback and when runtime->frame_bits is set by ALSA middle layer. When another substream is already running, if an interrupt is delivered during that window the irq handler calls pcm_pointer() which does a division by zero. The patch below makes the irq handler skip substreams that are initialized but not started yet. Cc to Clemens Ladisch because he proposed an alternate fix. For more information, please read the original thread in the linux-kernel mailing list: http://lkml.org/lkml/2010/2/2/187 Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1305f7c..641d7f0 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1821,7 +1821,9 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) /* The hardware doesn't tell us which substream caused the irq, thus we have to check all running substreams. */ for (ss = 0; ss < DSP_MAXPIPES; ss++) { - if ((substream = chip->substream[ss])) { + substream = chip->substream[ss]; + if (substream && ((struct audiopipe *)substream->runtime-> + private_data)->state == PIPE_STATE_STARTED) { period = pcm_pointer(substream) / substream->runtime->period_size; if (period != chip->last_period[ss]) { -- cgit v0.10.2 From e47c796d58a21fc58b00dffb7265bb66de987773 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 17 Feb 2010 09:49:54 +0200 Subject: ASoC: TWL4030: Use codec defaults for Headset initial configuration Disable the amplifiers for the headset outputs, and do not select routings by default to the headset outputs. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 277862e..6f5d4af 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -75,8 +75,8 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ 0x00, /* REG_EAR_CTL (0x21) */ - 0x24, /* REG_HS_SEL (0x22) */ - 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x00, /* REG_HS_SEL (0x22) */ + 0x00, /* REG_HS_GAIN_SET (0x23) */ 0x00, /* REG_HS_POPN_SET (0x24) */ 0x00, /* REG_PREDL_CTL (0x25) */ 0x00, /* REG_PREDR_CTL (0x26) */ -- cgit v0.10.2 From 6c5f1fed49f96a0600aa9a97ac3faf972c33a341 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 17 Feb 2010 14:30:44 +0000 Subject: ASoC: Make pmdown_time a long Fixes a warning. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/soc.h b/include/sound/soc.h index d9d88dd..5d234a8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -521,7 +521,7 @@ struct snd_soc_card { int (*set_bias_level)(struct snd_soc_card *, enum snd_soc_bias_level level); - int pmdown_time; + long pmdown_time; /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c2008bc..e1c0336 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -136,7 +136,7 @@ static ssize_t pmdown_time_show(struct device *dev, struct snd_soc_device *socdev = dev_get_drvdata(dev); struct snd_soc_card *card = socdev->card; - return sprintf(buf, "%d\n", card->pmdown_time); + return sprintf(buf, "%ld\n", card->pmdown_time); } static ssize_t pmdown_time_set(struct device *dev, -- cgit v0.10.2 From 7fb2d723e65cc793213515fa1da092b7c92a5b48 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Thu, 18 Feb 2010 07:01:20 +0100 Subject: ALSA: cs46xx - Do test writes to register AC97_REC_GAIN in snd_cs46xx_codec_reset() bypassing the register cache, so as to not clobber the cached register value during resume. Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 56fcf00..9fea5bb 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2266,7 +2266,7 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) return; /* test if we can write to the record gain volume register */ - snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x8a05); + snd_ac97_write(ac97, AC97_REC_GAIN, 0x8a05); if ((err = snd_ac97_read(ac97, AC97_REC_GAIN)) == 0x8a05) return; -- cgit v0.10.2 From 04510a74bfbcbfd53dd48b3094aad89d5eca1d28 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Thu, 18 Feb 2010 07:03:55 +0100 Subject: ALSA: cs46xx - fix some typos Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 9fea5bb..3f99a5e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2238,11 +2238,11 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) /* set the desired CODEC mode */ if (ac97->num == CS46XX_PRIMARY_CODEC_INDEX) { - snd_printdd("cs46xx: CODOEC1 mode %04x\n",0x0); - snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x0); + snd_printdd("cs46xx: CODEC1 mode %04x\n", 0x0); + snd_cs46xx_ac97_write(ac97, AC97_CSR_ACMODE, 0x0); } else if (ac97->num == CS46XX_SECONDARY_CODEC_INDEX) { - snd_printdd("cs46xx: CODOEC2 mode %04x\n",0x3); - snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x3); + snd_printdd("cs46xx: CODEC2 mode %04x\n", 0x3); + snd_cs46xx_ac97_write(ac97, AC97_CSR_ACMODE, 0x3); } else { snd_BUG(); /* should never happen ... */ } -- cgit v0.10.2 From ba579eb7b30791751f556ee01905636cda50c864 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sat, 20 Feb 2010 11:16:30 -0500 Subject: ALSA: hda: Use 3stack quirk for Toshiba Satellite L40-10Q BugLink: https://bugs.launchpad.net/bugs/524948 The OR has verified that the existing model=laptop-eapd quirk does not function correctly but instead needs model=3stack. Make this change so that manual corrections to module-init-tools file(s) are not required. Reported-by: Lasse Havelund CC: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 21011b5..7832f36 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1098,7 +1098,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), - SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50), -- cgit v0.10.2 From e458b1fadf9239d1fdb165ff4c4ea0d807041bec Mon Sep 17 00:00:00 2001 From: Luke Yelavich Date: Fri, 12 Feb 2010 16:28:29 +1100 Subject: ALSA: hda - Add Macmini 3,1 support BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989 Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The pinout is almost identical to the mb5 quirk, except for no microphone and the line-in mixer controls being on a different index. Everything works in 2ch mode, but as I am not sure what needs to be changed for 6ch mode, or whether the Mac Mini's chip supports 6ch mode, I have simply duplicated the code from the mb5 quirk for the mac mini chmode management. The new model parameter for this quirk is "macmini3". Signed-off-by: Luke Yelavich Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 0c7ebef..5efacf0 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -124,6 +124,7 @@ ALC882/883/885/888/889 asus-a7m ASUS A7M macpro MacPro support mb5 Macbook 5,1 + macmini3 Macmini 3,1 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection imac91 iMac 9,1 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0c22497..b5a6ba0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -211,6 +211,7 @@ enum { ALC885_MACPRO, ALC885_MBP3, ALC885_MB5, + ALC885_MACMINI3, ALC885_IMAC24, ALC885_IMAC91, ALC883_3ST_2ch_DIG, @@ -6751,6 +6752,14 @@ static struct hda_input_mux mb5_capture_source = { }, }; +static struct hda_input_mux macmini3_capture_source = { + .num_items = 2, + .items = { + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc883_3stack_6ch_intel = { .num_items = 4, .items = { @@ -6999,6 +7008,35 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; +/* + * 2ch + * Speakers/Woofer/HP = Front + * LineIn = Input + */ +static struct hda_verb alc885_macmini3_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } /* end */ +}; + +/* + * 6ch mode + * Speakers/HP = Front + * Woofer = LFE + * LineIn = Surround + */ +static struct hda_verb alc885_macmini3_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + { } /* end */ +}; + +static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { + { 2, alc885_mb5_ch2_init }, + { 6, alc885_mb5_ch6_init }, +}; + /* * 2ch mode @@ -7243,6 +7281,21 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc885_macmini3_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc885_imac91_mixer[] = { HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7617,6 +7670,53 @@ static struct hda_verb alc885_mb5_init_verbs[] = { { } }; +/* Macmini 3,1 */ +static struct hda_verb alc885_macmini3_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Line In pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } +}; + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7800,6 +7900,18 @@ static void alc885_mb5_automute(struct hda_codec *codec) } +static void alc885_macmini3_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + static void alc885_mb5_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -7808,6 +7920,14 @@ static void alc885_mb5_unsol_event(struct hda_codec *codec, alc885_mb5_automute(codec); } +static void alc885_macmini3_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -8974,6 +9094,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", + [ALC885_MACMINI3] = "macmini3", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9157,6 +9278,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), {} /* terminator */ }; @@ -9238,6 +9360,20 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc885_mb5_unsol_event, .init_hook = alc885_mb5_automute, }, + [ALC885_MACMINI3] = { + .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_macmini3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_macmini3_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes), + .input_mux = &macmini3_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_macmini3_unsol_event, + .init_hook = alc885_macmini3_automute, + }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, .init_verbs = { alc882_macpro_init_verbs }, -- cgit v0.10.2 From 9d54f08bc77bf6dfe835b945d03b6e127c9fc5a3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Feb 2010 08:34:40 +0100 Subject: ALSA: hda - Clean up Intel Mac unsol codes Use the standard unsol_event callback with each setup callback for IntelMac models with Realtek ALC885 codecs. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b5a6ba0..f8fb260 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7879,6 +7879,9 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x1a; } +#define alc885_mb5_setup alc885_imac24_setup +#define alc885_macmini3_setup alc885_imac24_setup + static void alc885_mbp3_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -7887,66 +7890,13 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } -static void alc885_mb5_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - -} - -static void alc885_macmini3_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -static void alc885_mb5_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_mb5_automute(codec); -} - -static void alc885_macmini3_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_mb5_automute(codec); -} - -static void alc885_imac91_automute(struct hda_codec *codec) +static void alc885_imac91_setup(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - -} + struct alc_spec *spec = codec->spec; -static void alc885_imac91_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_imac91_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[1] = 0x1a; } static struct hda_verb alc882_targa_verbs[] = { @@ -9357,8 +9307,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc885_mb5_unsol_event, - .init_hook = alc885_mb5_automute, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_mb5_setup, + .init_hook = alc_automute_amp, }, [ALC885_MACMINI3] = { .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, @@ -9371,8 +9322,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &macmini3_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc885_macmini3_unsol_event, - .init_hook = alc885_macmini3_automute, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_macmini3_setup, + .init_hook = alc_automute_amp, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, @@ -9411,8 +9363,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc885_imac91_unsol_event, - .init_hook = alc885_imac91_automute, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_imac91_setup, + .init_hook = alc_automute_amp, }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, -- cgit v0.10.2 From 2448158ed2ae64ef3219b51e0176a4e1151ba9ec Mon Sep 17 00:00:00 2001 From: Paul Menzel Date: Mon, 8 Feb 2010 20:37:26 +0100 Subject: ALSA: Typo. s/distrubs/disturbs/ Signed-off-by: Paul Menzel Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 06f230f..051cf51 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1411,7 +1411,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing * codec often screws up the controller chip, - * and distrubs the further communications. + * and disturbs the further communications. * Thus if an error occurs during probing, * better to reset the controller chip to * get back to the sanity state. -- cgit v0.10.2 From 0708cc582f0fe2578eaab722841caf2b4f8cfe37 Mon Sep 17 00:00:00 2001 From: Paul Menzel Date: Mon, 8 Feb 2010 20:42:46 +0100 Subject: ALSA: hda-intel: Add position_fix quirk for ASUS M2V-MX SE. With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1]. Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE. The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker. $ lspci -vvnn | grep -A10 Audio 20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASUSTeK Computer Inc. Device [1043:8290] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- SERR- Kernel driver in use: HDA Intel [1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24.camel%40mattotaupa&forum_name=alsa-user Signed-off-by: Paul Menzel Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 051cf51..22dcdc2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2264,6 +2264,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), {} -- cgit v0.10.2 From 40717382e0c1f572553e4fdefb489db4b95a5e7e Mon Sep 17 00:00:00 2001 From: Chris J Arges Date: Wed, 17 Feb 2010 12:12:52 -0600 Subject: ALSA: usbaudio Mbox support, output only Signed-off-by: Chris J Arges Signed-off-by: Takashi Iwai diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index e691eba..fc1d2cd 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2215,6 +2215,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Digidesign Mbox */ +{ + /* Thanks to Clemens Ladisch */ + USB_DEVICE(0x0dba, 0x1000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "MBox", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]){ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .format = SNDRV_PCM_FORMAT_S24_3BE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x02, + .ep_attr = 0x01, + .maxpacksize = 0x130, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = -1 + } + } + + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- cgit v0.10.2 From bf30a4309d4294d3eca248ea8a20c1c3570f5e74 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 22 Feb 2010 10:33:13 +0100 Subject: ALSA: via82xx: add quirk for D1289 motherboard Add a headphones-only quirk for the Fujitsu Siemens D1289. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Marc Haber Cc: Signed-off-by: Jaroslav Kysela diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8a332d2..03d6aea 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1791,6 +1791,12 @@ static struct ac97_quirk ac97_quirks[] = { .type = AC97_TUNE_HP_ONLY }, { + .subvendor = 0x110a, + .subdevice = 0x0079, + .name = "Fujitsu Siemens D1289", + .type = AC97_TUNE_HP_ONLY + }, + { .subvendor = 0x1019, .subdevice = 0x0a81, .name = "ECS K7VTA3", -- cgit v0.10.2 From b9dd94a87e5b4d0e864636698931aeeeb3c9d770 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 22 Feb 2010 13:27:13 +0200 Subject: ASoC: core: On resume also check the soc device state Check the card->codec on soc_resume to detect if the soc device is properly initialized. If the card->codec is NULL, than do not continue the resume operation, since the device is not initialized properly. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1c0336..a03bac9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -963,6 +963,12 @@ static int soc_resume(struct device *dev) struct snd_soc_card *card = socdev->card; struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (!card->codec) + return 0; + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume -- cgit v0.10.2 From d01aecdf900574cf6be7c1c6114e708801126baf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Feb 2010 08:07:15 +0100 Subject: ALSA: hda - Remove identical definitions for macmini3 model The channel mode definitions for macmini3 model are identical with mb5. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f8fb260..c74ca39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7008,35 +7008,7 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; -/* - * 2ch - * Speakers/Woofer/HP = Front - * LineIn = Input - */ -static struct hda_verb alc885_macmini3_ch2_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - { } /* end */ -}; - -/* - * 6ch mode - * Speakers/HP = Front - * Woofer = LFE - * LineIn = Surround - */ -static struct hda_verb alc885_macmini3_ch6_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - { } /* end */ -}; - -static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { - { 2, alc885_mb5_ch2_init }, - { 6, alc885_mb5_ch6_init }, -}; - +#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes /* * 2ch mode -- cgit v0.10.2 From 32679f95cac3b1bdf27dce8b5273e06af186fd91 Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Mon, 22 Feb 2010 17:31:09 -0800 Subject: ALSA: hda - enable snoop for Intel Cougar Point This patch enables snoop, eliminating static during playback. This patch supersedes the previous Cougar Point audio patch. Signed-off-by: Seth Heasley Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 22dcdc2..1adac8c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -448,6 +448,7 @@ struct azx { /* driver types */ enum { AZX_DRIVER_ICH, + AZX_DRIVER_PCH, AZX_DRIVER_SCH, AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, @@ -462,6 +463,7 @@ enum { static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ICH] = "HDA Intel", + [AZX_DRIVER_PCH] = "HDA Intel PCH", [AZX_DRIVER_SCH] = "HDA Intel MID", [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", @@ -1064,6 +1066,7 @@ static void azx_init_pci(struct azx *chip) 0x01, NVIDIA_HDA_ENABLE_COHBIT); break; case AZX_DRIVER_SCH: + case AZX_DRIVER_PCH: pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) { pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, @@ -2421,6 +2424,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, if (bdl_pos_adj[dev] < 0) { switch (chip->driver_type) { case AZX_DRIVER_ICH: + case AZX_DRIVER_PCH: bdl_pos_adj[dev] = 1; break; default: @@ -2700,7 +2704,7 @@ static struct pci_device_id azx_ids[] = { /* PCH */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, /* CPT */ - { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */ -- cgit v0.10.2 From 28e1b773083d349d5223f586a39fa30f5d0f1c36 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:09 +0100 Subject: ALSA: usbaudio: parse USB descriptors with structs In preparation of support for v2.0 audio class, use the structs from linux/usb/audio.h and add some new ones to describe the fields that are actually parsed by the descriptor decoders. Also, factor out code from usb_create_streams(). This makes it easier to adopt the new iteration logic needed for v2.0. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index eaf9dff..44f82d8 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -81,7 +81,7 @@ /* Terminal Control Selectors */ /* 4.3.2 Class-Specific AC Interface Descriptor */ -struct uac_ac_header_descriptor { +struct uac_ac_header_descriptor_v1 { __u8 bLength; /* 8 + n */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* UAC_MS_HEADER */ @@ -95,7 +95,7 @@ struct uac_ac_header_descriptor { /* As above, but more useful for defining your own descriptors: */ #define DECLARE_UAC_AC_HEADER_DESCRIPTOR(n) \ -struct uac_ac_header_descriptor_##n { \ +struct uac_ac_header_descriptor_v1_##n { \ __u8 bLength; \ __u8 bDescriptorType; \ __u8 bDescriptorSubtype; \ @@ -131,7 +131,7 @@ struct uac_input_terminal_descriptor { #define UAC_INPUT_TERMINAL_PROC_MICROPHONE_ARRAY 0x206 /* 4.3.2.2 Output Terminal Descriptor */ -struct uac_output_terminal_descriptor { +struct uac_output_terminal_descriptor_v1 { __u8 bLength; /* in bytes: 9 */ __u8 bDescriptorType; /* CS_INTERFACE descriptor type */ __u8 bDescriptorSubtype; /* OUTPUT_TERMINAL descriptor subtype */ @@ -171,7 +171,7 @@ struct uac_feature_unit_descriptor_##ch { \ } __attribute__ ((packed)) /* 4.5.2 Class-Specific AS Interface Descriptor */ -struct uac_as_header_descriptor { +struct uac_as_header_descriptor_v1 { __u8 bLength; /* in bytes: 7 */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* AS_GENERAL */ @@ -232,6 +232,19 @@ struct uac_format_type_i_discrete_descriptor_##n { \ #define UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(n) (8 + (n * 3)) +/* Formats - Audio Data Format Type I Codes */ + +struct uac_format_type_ii_discrete_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bFormatType; + __le16 wMaxBitRate; + __le16 wSamplesPerFrame; + __u8 bSamFreqType; + __u8 tSamFreq[][3]; +} __attribute__((packed)); + /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 #define UAC_FORMAT_TYPE_I 0x1 @@ -253,6 +266,17 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_FILL_MAX 0x80 /* A.10.2 Feature Unit Control Selectors */ + +struct uac_feature_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u8 bSourceID; + __u8 bControlSize; + __u8 controls[0]; /* variable length */ +} __attribute__((packed)); + #define UAC_FU_CONTROL_UNDEFINED 0x00 #define UAC_MUTE_CONTROL 0x01 #define UAC_VOLUME_CONTROL 0x02 diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c6b9c8c..f833dea 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -46,6 +46,8 @@ #include #include #include +#include + #include #include #include @@ -2421,15 +2423,17 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat * * @fmt: the format type descriptor */ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int pcm_format; int sample_width, sample_bytes; + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; /* FIXME: correct endianess and sign? */ pcm_format = -1; - sample_width = fmt[6]; - sample_bytes = fmt[5]; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubframeSize; + switch (format) { case 0: /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", @@ -2442,7 +2446,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor sample_width, sample_bytes); } /* check the format byte size */ - switch (fmt[5]) { + switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; break; @@ -2463,8 +2467,8 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor break; default: snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", - chip->dev->devnum, fp->iface, - fp->altsetting, sample_width, sample_bytes); + chip->dev->devnum, fp->iface, fp->altsetting, + sample_width, sample_bytes); break; } break; @@ -2564,11 +2568,12 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int pcm_format; + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - if (fmt[3] == USB_FORMAT_TYPE_III) { + if (fmt->bFormatType == USB_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -2590,23 +2595,27 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * if (pcm_format < 0) return -1; } + fp->format = pcm_format; - fp->channels = fmt[4]; + fp->channels = fmt->bNrChannels; + if (fp->channels < 1) { snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); return -1; } - return parse_audio_format_rates(chip, fp, fmt, 7); + return parse_audio_format_rates(chip, fp, fmt_raw, 7); } /* - * prase the format type II descriptor + * parse the format type II descriptor */ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int brate, framesize; + struct uac_format_type_ii_discrete_descriptor *fmt = fmt_raw; + switch (format) { case USB_AUDIO_FORMAT_AC3: /* FIXME: there is no AC3 format defined yet */ @@ -2622,20 +2631,25 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->format = SNDRV_PCM_FORMAT_MPEG; break; } + fp->channels = 1; - brate = combine_word(&fmt[4]); /* fmt[4,5] : wMaxBitRate (in kbps) */ - framesize = combine_word(&fmt[6]); /* fmt[6,7]: wSamplesPerFrame */ + + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); fp->frame_size = framesize; - return parse_audio_format_rates(chip, fp, fmt, 8); /* fmt[8..] sample rates */ + return parse_audio_format_rates(chip, fp, fmt_raw, 8); /* fmt[8..] sample rates */ } static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream) + int format, void *fmt_raw, int stream) { int err; + /* we only parse the common header of all format types here, + * so it is safe to take a type_i struct */ + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - switch (fmt[3]) { + switch (fmt->bFormatType) { case USB_FORMAT_TYPE_I: case USB_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt); @@ -2645,10 +2659,10 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); + chip->dev->devnum, fp->iface, fp->altsetting, fmt->bFormatType); return -1; } - fp->fmt_type = fmt[3]; + fp->fmt_type = fmt->bFormatType; if (err < 0) return err; #if 1 @@ -2659,7 +2673,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == USB_FORMAT_TYPE_I && + if (fmt->bFormatType == USB_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2708,6 +2722,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num = 4; for (i = 0; i < num; i++) { + struct uac_as_header_descriptor_v1 *as; + alts = &iface->altsetting[i]; altsd = get_iface_desc(alts); /* skip invalid one */ @@ -2726,7 +2742,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2003) && @@ -2734,20 +2750,21 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; /* get audio formats */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); - if (!fmt) { + as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } - if (fmt[0] < 7) { + if (as->bLength < sizeof(*as)) { snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } - format = (fmt[6] << 8) | fmt[5]; /* remember the format value */ + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ /* get format type */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); @@ -2875,6 +2892,65 @@ static void snd_usb_stream_disconnect(struct list_head *head) } } +static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int interface) +{ + struct usb_device *dev = chip->dev; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_interface *iface = usb_ifnum_to_if(dev, interface); + + if (!iface) { + snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + if (usb_interface_claimed(iface)) { + snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { + int err = snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL); + if (err < 0) { + snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + + return 0; + } + + if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { + snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", + dev->devnum, ctrlif, interface, altsd->bInterfaceClass); + /* skip non-supported classes */ + return -EINVAL; + } + + if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { + snd_printk(KERN_ERR "low speed audio streaming not supported\n"); + return -EINVAL; + } + + if (! parse_audio_endpoints(chip, interface)) { + usb_set_interface(dev, interface, 0); /* reset the current interface */ + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return -EINVAL; + } + + return 0; +} + /* * parse audio control descriptor and create pcm/midi streams */ @@ -2882,69 +2958,36 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) { struct usb_device *dev = chip->dev; struct usb_host_interface *host_iface; - struct usb_interface *iface; - unsigned char *p1; - int i, j; + struct uac_ac_header_descriptor_v1 *h1; + void *control_header; + int i; /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; - if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) { + control_header = snd_usb_find_csint_desc(host_iface->extra, + host_iface->extralen, + NULL, HEADER); + + if (!control_header) { snd_printk(KERN_ERR "cannot find HEADER\n"); return -EINVAL; } - if (! p1[7] || p1[0] < 8 + p1[7]) { - snd_printk(KERN_ERR "invalid HEADER\n"); + + h1 = control_header; + + if (!h1->bInCollection) { + snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); return -EINVAL; } - /* - * parse all USB audio streaming interfaces - */ - for (i = 0; i < p1[7]; i++) { - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - j = p1[8 + i]; - iface = usb_ifnum_to_if(dev, j); - if (!iface) { - snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", - dev->devnum, ctrlif, j); - continue; - } - if (usb_interface_claimed(iface)) { - snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, j); - continue; - } - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); - if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || - altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { - int err = snd_usbmidi_create(chip->card, iface, - &chip->midi_list, NULL); - if (err < 0) { - snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j); - continue; - } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - continue; - } - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && - altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { - snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, j, altsd->bInterfaceClass); - /* skip non-supported classes */ - continue; - } - if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { - snd_printk(KERN_ERR "low speed audio streaming not supported\n"); - continue; - } - if (! parse_audio_endpoints(chip, j)) { - usb_set_interface(dev, j, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - } + if (h1->bLength < sizeof(*h1) + h1->bInCollection) { + snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + return -EINVAL; } + for (i = 0; i < h1->bInCollection; i++) + snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + return 0; } @@ -3607,7 +3650,6 @@ static void *snd_usb_audio_probe(struct usb_device *dev, ifnum = get_iface_desc(alts)->bInterfaceNumber; id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); - if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) goto __err_val; diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 35b4830..11636a6 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -32,6 +32,8 @@ #include #include #include +#include + #include #include #include @@ -1086,29 +1088,30 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, * * most of controlls are defined here. */ -static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsigned char *ftr) +static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr) { int channels, i, j; struct usb_audio_term iterm; unsigned int master_bits, first_ch_bits; int err, csize; + struct uac_feature_unit_descriptor *ftr = _ftr; - if (ftr[0] < 7 || ! (csize = ftr[5]) || ftr[0] < 7 + csize) { + if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } /* parse the source unit */ - if ((err = parse_audio_unit(state, ftr[4])) < 0) + if ((err = parse_audio_unit(state, ftr->bSourceID)) < 0) return err; /* determine the input source type and name */ - if (check_input_term(state, ftr[4], &iterm) < 0) + if (check_input_term(state, ftr->bSourceID, &iterm) < 0) return -EINVAL; - channels = (ftr[0] - 7) / csize - 1; + channels = (ftr->bLength - 7) / csize - 1; - master_bits = snd_usb_combine_bytes(ftr + 6, csize); + master_bits = snd_usb_combine_bytes(ftr->controls, csize); /* master configuration quirks */ switch (state->chip->usb_id) { case USB_ID(0x08bb, 0x2702): @@ -1119,21 +1122,21 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig break; } if (channels > 0) - first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize); + first_ch_bits = snd_usb_combine_bytes(ftr->controls + csize, csize); else first_ch_bits = 0; /* check all control types */ for (i = 0; i < 10; i++) { unsigned int ch_bits = 0; for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(ftr + 6 + csize * (j+1), csize); + unsigned int mask = snd_usb_combine_bytes(ftr->controls + csize * (j+1), csize); if (mask & (1 << i)) ch_bits |= (1 << j); } if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, ftr, ch_bits, i, &iterm, unitid); + build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid); if (master_bits & (1 << i)) - build_feature_ctl(state, ftr, 0, i, &iterm, unitid); + build_feature_ctl(state, _ftr, 0, i, &iterm, unitid); } return 0; @@ -1780,7 +1783,7 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { - unsigned char *desc; + struct uac_output_terminal_descriptor_v1 *desc; struct mixer_build state; int err; const struct usbmix_ctl_map *map; @@ -1805,13 +1808,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) desc = NULL; while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) { - if (desc[0] < 9) + if (desc->bLength < 9) continue; /* invalid descriptor? */ - set_bit(desc[3], state.unitbitmap); /* mark terminal ID as visited */ - state.oterm.id = desc[3]; - state.oterm.type = combine_word(&desc[4]); - state.oterm.name = desc[8]; - err = parse_audio_unit(&state, desc[7]); + set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + state.oterm.id = desc->bTerminalID; + state.oterm.type = le16_to_cpu(desc->wTerminalType); + state.oterm.name = desc->iTerminal; + err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) return err; } -- cgit v0.10.2 From 8fee4aff8c89c229593b76a6ab172a9cad24b412 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:10 +0100 Subject: ALSA: usbaudio: introduce new types for audio class v2 This patch adds some definitions for audio class v2. Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have different numerical representations in both standards, so there is need for a _V1 add-on now. usbmixer.c is changed accordingly. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index 44f82d8..fb1a97b 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -25,6 +25,9 @@ #define USB_SUBCLASS_AUDIOSTREAMING 0x02 #define USB_SUBCLASS_MIDISTREAMING 0x03 +#define UAC_VERSION_1 0x00 +#define UAC_VERSION_2 0x20 + /* A.5 Audio Class-Specific AC Interface Descriptor Subtypes */ #define UAC_HEADER 0x01 #define UAC_INPUT_TERMINAL 0x02 @@ -180,6 +183,19 @@ struct uac_as_header_descriptor_v1 { __le16 wFormatTag; /* The Audio Data Format */ } __attribute__ ((packed)); +struct uac_as_header_descriptor_v2 { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalLink; + __u8 bmControls; + __u8 bFormatType; + __u32 bmFormats; + __u8 bNrChannels; + __u32 bmChannelConfig; + __u8 iChannelNames; +} __attribute__((packed)); + #define UAC_DT_AS_HEADER_SIZE 7 /* Formats - A.1.1 Audio Data Format Type I Codes */ @@ -232,6 +248,19 @@ struct uac_format_type_i_discrete_descriptor_##n { \ #define UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(n) (8 + (n * 3)) +struct uac_format_type_i_ext_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bSubslotSize; + __u8 bFormatType; + __u8 bBitResolution; + __u8 bHeaderLength; + __u8 bControlSize; + __u8 bSideBandProtocol; +} __attribute__((packed)); + + /* Formats - Audio Data Format Type I Codes */ struct uac_format_type_ii_discrete_descriptor { @@ -245,11 +274,26 @@ struct uac_format_type_ii_discrete_descriptor { __u8 tSamFreq[][3]; } __attribute__((packed)); +struct uac_format_type_ii_ext_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bFormatType; + __u16 wMaxBitRate; + __u16 wSamplesPerFrame; + __u8 bHeaderLength; + __u8 bSideBandProtocol; +} __attribute__((packed)); + + /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 #define UAC_FORMAT_TYPE_I 0x1 #define UAC_FORMAT_TYPE_II 0x2 #define UAC_FORMAT_TYPE_III 0x3 +#define UAC_EXT_FORMAT_TYPE_I 0x81 +#define UAC_EXT_FORMAT_TYPE_II 0x82 +#define UAC_EXT_FORMAT_TYPE_III 0x83 struct uac_iso_endpoint_descriptor { __u8 bLength; /* in bytes: 7 */ @@ -265,6 +309,19 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_PITCH_CONTROL 0x02 #define UAC_EP_CS_ATTR_FILL_MAX 0x80 +/* Audio class v2.0: CLOCK_SOURCE descriptor */ + +struct uac_clock_source_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bmAttributes; + __u8 bmControls; + __u8 bAssocTerminal; + __u8 iClockSource; +} __attribute__((packed)); + /* A.10.2 Feature Unit Control Selectors */ struct uac_feature_unit_descriptor { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9d8cea4..4f48293 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,8 +36,17 @@ #define MIXER_UNIT 0x04 #define SELECTOR_UNIT 0x05 #define FEATURE_UNIT 0x06 -#define PROCESSING_UNIT 0x07 -#define EXTENSION_UNIT 0x08 +#define PROCESSING_UNIT_V1 0x07 +#define EXTENSION_UNIT_V1 0x08 + +/* audio class v2 */ +#define EFFECT_UNIT 0x07 +#define PROCESSING_UNIT_V2 0x08 +#define EXTENSION_UNIT_V2 0x09 +#define CLOCK_SOURCE 0x0a +#define CLOCK_SELECTOR 0x0b +#define CLOCK_MULTIPLIER 0x0c +#define SAMPLE_RATE_CONVERTER 0x0d #define AS_GENERAL 0x01 #define FORMAT_TYPE 0x02 @@ -60,7 +69,7 @@ #define EP_CS_ATTR_PITCH_CONTROL 0x02 #define EP_CS_ATTR_FILL_MAX 0x80 -/* Audio Class specific Request Codes */ +/* Audio Class specific Request Codes (v1) */ #define SET_CUR 0x01 #define GET_CUR 0x81 @@ -74,6 +83,10 @@ #define GET_MEM 0x85 #define GET_STAT 0xff +/* Audio Class specific Request Codes (v2) */ +#define CS_CUR 0x01 +#define CS_RANGE 0x02 + /* Terminal Control Selectors */ #define COPY_PROTECT_CONTROL 0x01 diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 11636a6..ca79495 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -286,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT && p[3] == unit) + if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT_V1 && p[3] == unit) return p; } return NULL; @@ -607,9 +607,9 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm switch (iterm->type >> 16) { case SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case PROCESSING_UNIT: + case PROCESSING_UNIT_V1: strcpy(name, "Process Unit"); return 12; - case EXTENSION_UNIT: + case EXTENSION_UNIT_V1: strcpy(name, "Ext Unit"); return 8; case MIXER_UNIT: strcpy(name, "Mixer"); return 5; @@ -673,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->id = id; term->name = p1[9 + p1[0] - 1]; return 0; - case PROCESSING_UNIT: - case EXTENSION_UNIT: + case PROCESSING_UNIT_V1: + case EXTENSION_UNIT_V1: if (p1[6] == 1) { id = p1[7]; break; /* continue to parse */ @@ -1747,9 +1747,9 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_selector_unit(state, unitid, p1); case FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case PROCESSING_UNIT: + case PROCESSING_UNIT_V1: return parse_audio_processing_unit(state, unitid, p1); - case EXTENSION_UNIT: + case EXTENSION_UNIT_V1: return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); -- cgit v0.10.2 From 53ee98fe8ac77d00bacc1c814d450d83cbd193d4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:11 +0100 Subject: ALSA: usbaudio: implement basic set of class v2.0 parser This adds a number of parsers for audio class v2.0. In particular, the following internals are different and now handled by the code: * the number of streaming interfaces is now reported by an interface association descriptor. The old approach using a proprietary descriptor is deprecated. * The number of channels per interface is now stored in the AS_GENERAL descriptor (used to be part of the FORMAT_TYPE descriptor). * The list of supported sample rates is no longer stored in a variable length appendix of the format_type descriptor but is retrieved from the device using a class specific GET_RANGE command. * Supported sample formats are now reported as 32bit bitmap rather than a fixed value. For now, this is worked around by choosing just one of them. * A devices needs to have at least one CLOCK_SOURCE descriptor which denotes a clockID that is needed im the class request command. * Many descriptors (format_type, ...) have changed their layout. Handle this by casting the descriptors to the appropriate structs. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index f833dea..411a6cf 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2422,17 +2422,53 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat * * @format: the format tag (wFormatTag) * @fmt: the format type descriptor */ -static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_i_type(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + int protocol) { - int pcm_format; + int pcm_format, i; int sample_width, sample_bytes; - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; + + switch (protocol) { + case UAC_VERSION_1: { + struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubframeSize; + break; + } + + case UAC_VERSION_2: { + struct uac_format_type_i_ext_descriptor *fmt = _fmt; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubslotSize; + + /* + * FIXME + * USB audio class v2 devices specify a bitmap of possible + * audio formats rather than one fix value. For now, we just + * pick one of them and report that as the only possible + * value for this setting. + * The bit allocation map is in fact compatible to the + * wFormatTag of the v1 AS streaming descriptors, which is why + * we can simply map the matrix. + */ + + for (i = 0; i < 5; i++) + if (format & (1UL << i)) { + format = i + 1; + break; + } + + break; + } + + default: + return -EINVAL; + } /* FIXME: correct endianess and sign? */ pcm_format = -1; - sample_width = fmt->bBitResolution; - sample_bytes = fmt->bSubframeSize; switch (format) { case 0: /* some devices don't define this correctly... */ @@ -2446,6 +2482,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor sample_width, sample_bytes); } /* check the format byte size */ + printk(" XXXXX SAMPLE BYTES %d\n", sample_bytes); switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; @@ -2500,7 +2537,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor /* * parse the format descriptor and stores the possible sample rates - * on the audioformat table. + * on the audioformat table (audio class v1). * * @dev: usb device * @fp: audioformat record @@ -2508,8 +2545,8 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor * @offset: the start offset of descriptor pointing the rate type * (7 for type I and II, 8 for type II) */ -static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioformat *fp, - unsigned char *fmt, int offset) +static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audioformat *fp, + unsigned char *fmt, int offset) { int nr_rates = fmt[offset]; @@ -2565,13 +2602,85 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform } /* + * parse the format descriptor and stores the possible sample rates + * on the audioformat table (audio class v2). + */ +static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, + struct audioformat *fp, + struct usb_host_interface *iface) +{ + struct usb_device *dev = chip->dev; + unsigned char tmp[2], *data; + int i, nr_rates, data_size, ret = 0; + + /* get the number of sample rates first by only fetching 2 bytes */ + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000); + + if (ret < 0) { + snd_printk(KERN_ERR "unable to retrieve number of sample rates\n"); + goto err; + } + + nr_rates = (tmp[1] << 8) | tmp[0]; + data_size = 2 + 12 * nr_rates; + data = kzalloc(data_size, GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto err; + } + + /* now get the full information */ + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + 0x0100, chip->clock_id << 8, data, data_size, 1000); + + if (ret < 0) { + snd_printk(KERN_ERR "unable to retrieve sample rate range\n"); + ret = -EINVAL; + goto err_free; + } + + fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); + if (!fp->rate_table) { + ret = -ENOMEM; + goto err_free; + } + + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; + + for (i = 0; i < nr_rates; i++) { + int rate = combine_quad(&data[2 + 12 * i]); + + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) + fp->rate_min = rate; + if (!fp->rate_max || rate > fp->rate_max) + fp->rate_max = rate; + fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; + } + +err_free: + kfree(data); +err: + return ret; +} + +/* * parse the format type I and III descriptors */ -static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_i(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + struct usb_host_interface *iface) { - int pcm_format; - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; + struct usb_interface_descriptor *altsd = get_iface_desc(iface); + struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + int protocol = altsd->bInterfaceProtocol; + int pcm_format, ret; if (fmt->bFormatType == USB_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx @@ -2591,30 +2700,49 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * pcm_format = SNDRV_PCM_FORMAT_S16_LE; } } else { - pcm_format = parse_audio_format_i_type(chip, fp, format, fmt); + pcm_format = parse_audio_format_i_type(chip, fp, format, fmt, protocol); if (pcm_format < 0) return -1; } fp->format = pcm_format; - fp->channels = fmt->bNrChannels; + + /* gather possible sample rates */ + /* audio class v1 reports possible sample rates as part of the + * proprietary class specific descriptor. + * audio class v2 uses class specific EP0 range requests for that. + */ + switch (protocol) { + case UAC_VERSION_1: + fp->channels = fmt->bNrChannels; + ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7); + break; + case UAC_VERSION_2: + /* fp->channels is already set in this case */ + ret = parse_audio_format_rates_v2(chip, fp, iface); + break; + } if (fp->channels < 1) { snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); return -1; } - return parse_audio_format_rates(chip, fp, fmt_raw, 7); + + return ret; } /* * parse the format type II descriptor */ -static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_ii(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + struct usb_host_interface *iface) { - int brate, framesize; - struct uac_format_type_ii_discrete_descriptor *fmt = fmt_raw; + int brate, framesize, ret; + struct usb_interface_descriptor *altsd = get_iface_desc(iface); + int protocol = altsd->bInterfaceProtocol; switch (format) { case USB_AUDIO_FORMAT_AC3: @@ -2634,35 +2762,50 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->channels = 1; - brate = le16_to_cpu(fmt->wMaxBitRate); - framesize = le16_to_cpu(fmt->wSamplesPerFrame); - snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); - fp->frame_size = framesize; - return parse_audio_format_rates(chip, fp, fmt_raw, 8); /* fmt[8..] sample rates */ + switch (protocol) { + case UAC_VERSION_1: { + struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); + snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); + fp->frame_size = framesize; + ret = parse_audio_format_rates_v1(chip, fp, _fmt, 8); /* fmt[8..] sample rates */ + break; + } + case UAC_VERSION_2: { + struct uac_format_type_ii_ext_descriptor *fmt = _fmt; + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); + snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); + fp->frame_size = framesize; + ret = parse_audio_format_rates_v2(chip, fp, iface); + break; + } + } + + return ret; } static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw, int stream) + int format, unsigned char *fmt, int stream, + struct usb_host_interface *iface) { int err; - /* we only parse the common header of all format types here, - * so it is safe to take a type_i struct */ - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - switch (fmt->bFormatType) { + switch (fmt[3]) { case USB_FORMAT_TYPE_I: case USB_FORMAT_TYPE_III: - err = parse_audio_format_i(chip, fp, format, fmt); + err = parse_audio_format_i(chip, fp, format, fmt, iface); break; case USB_FORMAT_TYPE_II: - err = parse_audio_format_ii(chip, fp, format, fmt); + err = parse_audio_format_ii(chip, fp, format, fmt, iface); break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt->bFormatType); + chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); return -1; } - fp->fmt_type = fmt->bFormatType; + fp->fmt_type = fmt[3]; if (err < 0) return err; #if 1 @@ -2673,7 +2816,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt->bFormatType == USB_FORMAT_TYPE_I && + if (fmt[3] == USB_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2702,10 +2845,10 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; int i, altno, err, stream; - int format; + int format = 0, num_channels = 0; struct audioformat *fp = NULL; unsigned char *fmt, *csep; - int num; + int num, protocol; dev = chip->dev; @@ -2722,10 +2865,9 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num = 4; for (i = 0; i < num; i++) { - struct uac_as_header_descriptor_v1 *as; - alts = &iface->altsetting[i]; altsd = get_iface_desc(alts); + protocol = altsd->bInterfaceProtocol; /* skip invalid one */ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || @@ -2742,7 +2884,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2003) && @@ -2750,21 +2892,54 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; /* get audio formats */ - as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + switch (protocol) { + case UAC_VERSION_1: { + struct uac_as_header_descriptor_v1 *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + break; } - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; + case UAC_VERSION_2: { + struct uac_as_header_descriptor_v2 *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + num_channels = as->bNrChannels; + format = le32_to_cpu(as->bmFormats); + + break; } - format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + default: + snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", + dev->devnum, iface_no, altno, protocol); + continue; + } /* get format type */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); @@ -2773,7 +2948,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno); continue; } - if (fmt[0] < 8) { + if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || + ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; @@ -2787,6 +2963,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && fp->format == SNDRV_PCM_FORMAT_S16_LE && + protocol == UAC_VERSION_1 && le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == fp->maxpacksize * 2) continue; @@ -2815,6 +2992,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + /* num_channels is only set for v2 interfaces */ + fp->channels = num_channels; if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) * (fp->maxpacksize & 0x7ff); @@ -2850,7 +3029,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } /* ok, let's parse further... */ - if (parse_audio_format(chip, fp, format, fmt, stream) < 0) { + if (parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { kfree(fp->rate_table); kfree(fp); continue; @@ -2958,35 +3137,82 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) { struct usb_device *dev = chip->dev; struct usb_host_interface *host_iface; - struct uac_ac_header_descriptor_v1 *h1; + struct usb_interface_descriptor *altsd; void *control_header; - int i; + int i, protocol; /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; control_header = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER); + altsd = get_iface_desc(host_iface); + protocol = altsd->bInterfaceProtocol; if (!control_header) { snd_printk(KERN_ERR "cannot find HEADER\n"); return -EINVAL; } - h1 = control_header; + switch (protocol) { + case UAC_VERSION_1: { + struct uac_ac_header_descriptor_v1 *h1 = control_header; - if (!h1->bInCollection) { - snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); - return -EINVAL; + if (!h1->bInCollection) { + snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); + return -EINVAL; + } + + if (h1->bLength < sizeof(*h1) + h1->bInCollection) { + snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + return -EINVAL; + } + + for (i = 0; i < h1->bInCollection; i++) + snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + + break; } - if (h1->bLength < sizeof(*h1) + h1->bInCollection) { - snd_printk(KERN_ERR "invalid HEADER (v1)\n"); - return -EINVAL; + case UAC_VERSION_2: { + struct uac_clock_source_descriptor *cs; + struct usb_interface_assoc_descriptor *assoc = + usb_ifnum_to_if(dev, ctrlif)->intf_assoc; + + if (!assoc) { + snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n"); + return -EINVAL; + } + + /* FIXME: for now, we expect there is at least one clock source + * descriptor and we always take the first one. + * We should properly support devices with multiple clock sources, + * clock selectors and sample rate conversion units. */ + + cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, + NULL, CLOCK_SOURCE); + + if (!cs) { + snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); + return -EINVAL; + } + + chip->clock_id = cs->bClockID; + + for (i = 0; i < assoc->bInterfaceCount; i++) { + int intf = assoc->bFirstInterface + i; + + if (intf != ctrlif) + snd_usb_create_stream(chip, ctrlif, intf); + } + + break; } - for (i = 0; i < h1->bInCollection; i++) - snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + default: + snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); + return -EINVAL; + } return 0; } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 4f48293..26daf68 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -142,6 +142,9 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + /* for audio class v2 */ + int clock_id; + struct list_head pcm_list; /* list of pcm streams */ int pcm_devs; -- cgit v0.10.2 From 7b8a043f2686af9f41e313a78ed5e98233e5fded Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:12 +0100 Subject: ALSA: usbmixer: bail out early when parsing audio class v2 descriptors This is just a quick hack that needs to be removed once the new units defined by the audio class v2.0 standard are supported. However, it allows using these devices for now, without mixer support. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index ca79495..42bb95c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -2258,7 +2258,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, }; struct usb_mixer_interface *mixer; struct snd_info_entry *entry; - int err; + struct usb_host_interface *host_iface; + int err, protocol; strcpy(chip->card->mixername, "USB Mixer"); @@ -2275,6 +2276,16 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, return -ENOMEM; } + host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; + protocol = host_iface->desc.bInterfaceProtocol; + + /* FIXME! */ + if (protocol != UAC_VERSION_1) { + snd_printk(KERN_WARNING "mixer interface protocol 0x%02x not yet supported\n", + protocol); + return 0; + } + if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) goto _error; -- cgit v0.10.2 From de48c7bc6f93c6c8e0be8612c9d72a2dc92eaa01 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:13 +0100 Subject: ALSA: usbaudio: consolidate header files Use the definitions from linux/usb/audio.h all over the ALSA USB audio driver and add some missing definitions there as well. Use the endpoint attribute macros from linux/usb/ch9 and remove the own things from sound/usb/usbaudio.h. Now things are also nicely prefixed which makes understanding the code easier. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index fb1a97b..6bb2936 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -35,8 +35,17 @@ #define UAC_MIXER_UNIT 0x04 #define UAC_SELECTOR_UNIT 0x05 #define UAC_FEATURE_UNIT 0x06 -#define UAC_PROCESSING_UNIT 0x07 -#define UAC_EXTENSION_UNIT 0x08 +#define UAC_PROCESSING_UNIT_V1 0x07 +#define UAC_EXTENSION_UNIT_V1 0x08 + +/* UAC v2.0 types */ +#define UAC_EFFECT_UNIT 0x07 +#define UAC_PROCESSING_UNIT_V2 0x08 +#define UAC_EXTENSION_UNIT_V2 0x09 +#define UAC_CLOCK_SOURCE 0x0a +#define UAC_CLOCK_SELECTOR 0x0b +#define UAC_CLOCK_MULTIPLIER 0x0c +#define UAC_SAMPLE_RATE_CONVERTER 0x0d /* A.6 Audio Class-Specific AS Interface Descriptor Subtypes */ #define UAC_AS_GENERAL 0x01 @@ -69,6 +78,10 @@ #define UAC_GET_STAT 0xff +/* Audio class v2.0 handles all the parameter calls differently */ +#define UAC2_CS_CUR 0x01 +#define UAC2_CS_RANGE 0x02 + /* MIDI - A.1 MS Class-Specific Interface Descriptor Subtypes */ #define UAC_MS_HEADER 0x01 #define UAC_MIDI_IN_JACK 0x02 @@ -133,6 +146,10 @@ struct uac_input_terminal_descriptor { #define UAC_INPUT_TERMINAL_MICROPHONE_ARRAY 0x205 #define UAC_INPUT_TERMINAL_PROC_MICROPHONE_ARRAY 0x206 +/* Terminals - control selectors */ + +#define UAC_TERMINAL_CS_COPY_PROTECT_CONTROL 0x01 + /* 4.3.2.2 Output Terminal Descriptor */ struct uac_output_terminal_descriptor_v1 { __u8 bLength; /* in bytes: 9 */ @@ -263,6 +280,9 @@ struct uac_format_type_i_ext_descriptor { /* Formats - Audio Data Format Type I Codes */ +#define UAC_FORMAT_TYPE_II_MPEG 0x1001 +#define UAC_FORMAT_TYPE_II_AC3 0x1002 + struct uac_format_type_ii_discrete_descriptor { __u8 bLength; __u8 bDescriptorType; @@ -285,6 +305,13 @@ struct uac_format_type_ii_ext_descriptor { __u8 bSideBandProtocol; } __attribute__((packed)); +/* type III */ +#define UAC_FORMAT_TYPE_III_IEC1937_AC3 0x2001 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG1_LAYER1 0x2002 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_NOEXT 0x2003 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_EXT 0x2004 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_LAYER1_LS 0x2005 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_LAYER23_LS 0x2006 /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 411a6cf..c539f7f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -47,6 +47,7 @@ #include #include #include +#include #include #include @@ -598,7 +599,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; - if (subs->fmt_type == USB_FORMAT_TYPE_II) { + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not * supported yet */ @@ -1106,7 +1107,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri u->packets = (i + 1) * total_packs / subs->nurbs - i * total_packs / subs->nurbs; u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == USB_FORMAT_TYPE_II) + if (subs->fmt_type == UAC_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); if (!u->urb) @@ -1182,7 +1183,7 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned if (i >= fp->nr_rates) continue; } - attr = fp->ep_attr & EP_ATTR_MASK; + attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; if (! found) { found = fp; cur_attr = attr; @@ -1194,14 +1195,14 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned * M-audio audiophile USB. */ if (attr != cur_attr) { - if ((attr == EP_ATTR_ASYNC && + if ((attr == USB_ENDPOINT_SYNC_ASYNC && subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (attr == EP_ATTR_ADAPTIVE && + (attr == USB_ENDPOINT_SYNC_ADAPTIVE && subs->direction == SNDRV_PCM_STREAM_CAPTURE)) continue; - if ((cur_attr == EP_ATTR_ASYNC && + if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC && subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (cur_attr == EP_ATTR_ADAPTIVE && + (cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE && subs->direction == SNDRV_PCM_STREAM_CAPTURE)) { found = fp; cur_attr = attr; @@ -1231,11 +1232,11 @@ static int init_usb_pitch(struct usb_device *dev, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint has pitch control, enable it */ - if (fmt->attributes & EP_CS_ATTR_PITCH_CONTROL) { + if (fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL) { data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { + UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n", dev->devnum, iface, ep); return err; @@ -1254,21 +1255,21 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint has sampling rate control, set it */ - if (fmt->attributes & EP_CS_ATTR_SAMPLE_RATE) { + if (fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE) { int crate; data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ @@ -1386,9 +1387,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) * descriptors which fool us. if it has only one EP, * assume it as adaptive-out or sync-in. */ - attr = fmt->ep_attr & EP_ATTR_MASK; - if (((is_playback && attr == EP_ATTR_ASYNC) || - (! is_playback && attr == EP_ATTR_ADAPTIVE)) && + attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || + (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && altsd->bNumEndpoints >= 2) { /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress @@ -1428,7 +1429,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } /* always fill max packet size */ - if (fmt->attributes & EP_CS_ATTR_FILL_MAX) + if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) subs->fill_max = 1; if ((err = init_usb_pitch(dev, subs->interface, alts, fmt)) < 0) @@ -1886,7 +1887,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.channels_min = fp->channels; if (runtime->hw.channels_max < fp->channels) runtime->hw.channels_max = fp->channels; - if (fp->fmt_type == USB_FORMAT_TYPE_II && fp->frame_size > 0) { + if (fp->fmt_type == UAC_FORMAT_TYPE_II && fp->frame_size > 0) { /* FIXME: there might be more than one audio formats... */ runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = fp->frame_size; @@ -2120,7 +2121,7 @@ static struct usb_device_id usb_audio_ids [] = { #include "usbquirks.h" { .match_flags = (USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS), .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { } /* Terminating entry */ }; @@ -2159,7 +2160,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, fp->endpoint & USB_DIR_IN ? "IN" : "OUT", - sync_types[(fp->ep_attr & EP_ATTR_MASK) >> 2]); + sync_types[(fp->ep_attr & USB_ENDPOINT_SYNCTYPE) >> 2]); if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) { snd_iprintf(buffer, " Rates: %d - %d (continuous)\n", fp->rate_min, fp->rate_max); @@ -2471,11 +2472,11 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, pcm_format = -1; switch (format) { - case 0: /* some devices don't define this correctly... */ + case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", chip->dev->devnum, fp->iface, fp->altsetting); /* fall-through */ - case USB_AUDIO_FORMAT_PCM: + case UAC_FORMAT_TYPE_I_PCM: if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, @@ -2509,7 +2510,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, break; } break; - case USB_AUDIO_FORMAT_PCM8: + case UAC_FORMAT_TYPE_I_PCM8: pcm_format = SNDRV_PCM_FORMAT_U8; /* Dallas DS4201 workaround: it advertises U8 format, but really @@ -2517,13 +2518,13 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x04fa, 0x4201)) pcm_format = SNDRV_PCM_FORMAT_S8; break; - case USB_AUDIO_FORMAT_IEEE_FLOAT: + case UAC_FORMAT_TYPE_I_IEEE_FLOAT: pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; break; - case USB_AUDIO_FORMAT_ALAW: + case UAC_FORMAT_TYPE_I_ALAW: pcm_format = SNDRV_PCM_FORMAT_A_LAW; break; - case USB_AUDIO_FORMAT_MU_LAW: + case UAC_FORMAT_TYPE_I_MULAW: pcm_format = SNDRV_PCM_FORMAT_MU_LAW; break; default: @@ -2551,7 +2552,7 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof int nr_rates = fmt[offset]; if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", chip->dev->devnum, fp->iface, fp->altsetting); return -1; } @@ -2614,7 +2615,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, int i, nr_rates, data_size, ret = 0; /* get the number of sample rates first by only fetching 2 bytes */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000); @@ -2632,7 +2633,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, } /* now get the full information */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, 0x0100, chip->clock_id << 8, data, data_size, 1000); @@ -2682,7 +2683,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, int protocol = altsd->bInterfaceProtocol; int pcm_format, ret; - if (fmt->bFormatType == USB_FORMAT_TYPE_III) { + if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -2745,12 +2746,12 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, int protocol = altsd->bInterfaceProtocol; switch (format) { - case USB_AUDIO_FORMAT_AC3: + case UAC_FORMAT_TYPE_II_AC3: /* FIXME: there is no AC3 format defined yet */ // fp->format = SNDRV_PCM_FORMAT_AC3; fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */ break; - case USB_AUDIO_FORMAT_MPEG: + case UAC_FORMAT_TYPE_II_MPEG: fp->format = SNDRV_PCM_FORMAT_MPEG; break; default: @@ -2793,11 +2794,11 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp int err; switch (fmt[3]) { - case USB_FORMAT_TYPE_I: - case USB_FORMAT_TYPE_III: + case UAC_FORMAT_TYPE_I: + case UAC_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt, iface); break; - case USB_FORMAT_TYPE_II: + case UAC_FORMAT_TYPE_II: err = parse_audio_format_ii(chip, fp, format, fmt, iface); break; default: @@ -2816,7 +2817,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == USB_FORMAT_TYPE_I && + if (fmt[3] == UAC_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2871,7 +2872,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* skip invalid one */ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING && + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || altsd->bNumEndpoints < 1 || le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) @@ -2895,16 +2896,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) switch (protocol) { case UAC_VERSION_1: { struct uac_as_header_descriptor_v1 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } @@ -2915,16 +2916,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) case UAC_VERSION_2: { struct uac_as_header_descriptor_v2 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } @@ -2942,15 +2943,15 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); if (!fmt) { - snd_printk(KERN_ERR "%d:%u:%d : no FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; } if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { - snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; } @@ -2972,7 +2973,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* Creamware Noah has this descriptor after the 2nd endpoint */ if (!csep && altsd->bNumEndpoints >= 2) csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); - if (!csep || csep[0] < 7 || csep[2] != EP_GENERAL) { + if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) { snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" " class specific endpoint descriptor\n", dev->devnum, iface_no, altno); @@ -3006,12 +3007,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* Optoplay sets the sample rate attribute although * it seems not supporting it in fact. */ - fp->attributes &= ~EP_CS_ATTR_SAMPLE_RATE; + fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ /* doesn't set the sample rate attribute, but supports it */ - fp->attributes |= EP_CS_ATTR_SAMPLE_RATE; + fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is @@ -3020,11 +3021,11 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) * plantronics headset and Griffin iMic have set adaptive-in * although it's really not... */ - fp->ep_attr &= ~EP_ATTR_MASK; + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; if (stream == SNDRV_PCM_STREAM_PLAYBACK) - fp->ep_attr |= EP_ATTR_ADAPTIVE; + fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; else - fp->ep_attr |= EP_ATTR_SYNC; + fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; break; } @@ -3094,7 +3095,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int altsd = get_iface_desc(alts); if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { int err = snd_usbmidi_create(chip->card, iface, &chip->midi_list, NULL); if (err < 0) { @@ -3109,7 +3110,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING) { snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, interface, altsd->bInterfaceClass); /* skip non-supported classes */ @@ -3145,12 +3146,12 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; control_header = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, HEADER); + NULL, UAC_HEADER); altsd = get_iface_desc(host_iface); protocol = altsd->bInterfaceProtocol; if (!control_header) { - snd_printk(KERN_ERR "cannot find HEADER\n"); + snd_printk(KERN_ERR "cannot find UAC_HEADER\n"); return -EINVAL; } @@ -3164,7 +3165,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } if (h1->bLength < sizeof(*h1) + h1->bInCollection) { - snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + snd_printk(KERN_ERR "invalid UAC_HEADER (v1)\n"); return -EINVAL; } @@ -3190,7 +3191,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) * clock selectors and sample rate conversion units. */ cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, CLOCK_SOURCE); + NULL, UAC_CLOCK_SOURCE); if (!cs) { snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); @@ -3302,7 +3303,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, - .fmt_type = USB_FORMAT_TYPE_I, + .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -3394,7 +3395,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, { static const struct audioformat ua1000_format = { .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, + .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, .attributes = 0, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 26daf68..6b016d4 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -21,106 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ - -/* - */ - -#define USB_SUBCLASS_AUDIO_CONTROL 0x01 -#define USB_SUBCLASS_AUDIO_STREAMING 0x02 -#define USB_SUBCLASS_MIDI_STREAMING 0x03 -#define USB_SUBCLASS_VENDOR_SPEC 0xff - -#define HEADER 0x01 -#define INPUT_TERMINAL 0x02 -#define OUTPUT_TERMINAL 0x03 -#define MIXER_UNIT 0x04 -#define SELECTOR_UNIT 0x05 -#define FEATURE_UNIT 0x06 -#define PROCESSING_UNIT_V1 0x07 -#define EXTENSION_UNIT_V1 0x08 - -/* audio class v2 */ -#define EFFECT_UNIT 0x07 -#define PROCESSING_UNIT_V2 0x08 -#define EXTENSION_UNIT_V2 0x09 -#define CLOCK_SOURCE 0x0a -#define CLOCK_SELECTOR 0x0b -#define CLOCK_MULTIPLIER 0x0c -#define SAMPLE_RATE_CONVERTER 0x0d - -#define AS_GENERAL 0x01 -#define FORMAT_TYPE 0x02 -#define FORMAT_SPECIFIC 0x03 - -#define EP_GENERAL 0x01 - -#define MS_GENERAL 0x01 -#define MIDI_IN_JACK 0x02 -#define MIDI_OUT_JACK 0x03 - -/* endpoint attributes */ -#define EP_ATTR_MASK 0x0c -#define EP_ATTR_ASYNC 0x04 -#define EP_ATTR_ADAPTIVE 0x08 -#define EP_ATTR_SYNC 0x0c - -/* cs endpoint attributes */ -#define EP_CS_ATTR_SAMPLE_RATE 0x01 -#define EP_CS_ATTR_PITCH_CONTROL 0x02 -#define EP_CS_ATTR_FILL_MAX 0x80 - -/* Audio Class specific Request Codes (v1) */ - -#define SET_CUR 0x01 -#define GET_CUR 0x81 -#define SET_MIN 0x02 -#define GET_MIN 0x82 -#define SET_MAX 0x03 -#define GET_MAX 0x83 -#define SET_RES 0x04 -#define GET_RES 0x84 -#define SET_MEM 0x05 -#define GET_MEM 0x85 -#define GET_STAT 0xff - -/* Audio Class specific Request Codes (v2) */ -#define CS_CUR 0x01 -#define CS_RANGE 0x02 - -/* Terminal Control Selectors */ - -#define COPY_PROTECT_CONTROL 0x01 - -/* Endpoint Control Selectors */ - -#define SAMPLING_FREQ_CONTROL 0x01 -#define PITCH_CONTROL 0x02 - -/* Format Types */ -#define USB_FORMAT_TYPE_I 0x01 -#define USB_FORMAT_TYPE_II 0x02 -#define USB_FORMAT_TYPE_III 0x03 - -/* type I */ -#define USB_AUDIO_FORMAT_PCM 0x01 -#define USB_AUDIO_FORMAT_PCM8 0x02 -#define USB_AUDIO_FORMAT_IEEE_FLOAT 0x03 -#define USB_AUDIO_FORMAT_ALAW 0x04 -#define USB_AUDIO_FORMAT_MU_LAW 0x05 - -/* type II */ -#define USB_AUDIO_FORMAT_MPEG 0x1001 -#define USB_AUDIO_FORMAT_AC3 0x1002 - -/* type III */ -#define USB_AUDIO_FORMAT_IEC1937_AC3 0x2001 -#define USB_AUDIO_FORMAT_IEC1937_MPEG1_LAYER1 0x2002 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_NOEXT 0x2003 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_EXT 0x2004 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER1_LS 0x2005 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER23_LS 0x2006 - - /* maximum number of endpoints per interface */ #define MIDI_MAX_ENDPOINTS 2 diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index b2da478..2c59afd 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -46,6 +46,8 @@ #include #include #include +#include + #include #include #include @@ -1540,7 +1542,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, if (hostif->extralen >= 7 && ms_header->bLength >= 7 && ms_header->bDescriptorType == USB_DT_CS_INTERFACE && - ms_header->bDescriptorSubtype == HEADER) + ms_header->bDescriptorSubtype == UAC_HEADER) snd_printdd(KERN_INFO "MIDIStreaming version %02x.%02x\n", ms_header->bcdMSC[1], ms_header->bcdMSC[0]); else @@ -1556,7 +1558,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, if (hostep->extralen < 4 || ms_ep->bLength < 4 || ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != MS_GENERAL) + ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { @@ -1768,9 +1770,9 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2; cs_desc += cs_desc[0]) { if (cs_desc[1] == USB_DT_CS_INTERFACE) { - if (cs_desc[2] == MIDI_IN_JACK) + if (cs_desc[2] == UAC_MIDI_IN_JACK) endpoint->in_cables = (endpoint->in_cables << 1) | 1; - else if (cs_desc[2] == MIDI_OUT_JACK) + else if (cs_desc[2] == UAC_MIDI_OUT_JACK) endpoint->out_cables = (endpoint->out_cables << 1) | 1; } } diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 42bb95c..8e8f871b 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -286,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT_V1 && p[3] == unit) + if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC_EXTENSION_UNIT_V1 && p[3] == unit) return p; } return NULL; @@ -407,14 +407,14 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value) { - return get_ctl_value(cval, GET_CUR, validx, value); + return get_ctl_value(cval, UAC_GET_CUR, validx, value); } /* channel = 0: master, 1 = first channel */ static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, int channel, int *value) { - return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value); + return get_ctl_value(cval, UAC_GET_CUR, (cval->control << 8) | channel, value); } static int get_cur_mix_value(struct usb_mixer_elem_info *cval, @@ -468,14 +468,14 @@ static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value) { - return set_ctl_value(cval, SET_CUR, validx, value); + return set_ctl_value(cval, UAC_SET_CUR, validx, value); } static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value) { int err; - err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, + err = set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, value); if (err < 0) return err; @@ -605,13 +605,13 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm if (term_only) return 0; switch (iterm->type >> 16) { - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case PROCESSING_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: strcpy(name, "Process Unit"); return 12; - case EXTENSION_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: strcpy(name, "Ext Unit"); return 8; - case MIXER_UNIT: + case UAC_MIXER_UNIT: strcpy(name, "Mixer"); return 5; default: return sprintf(name, "Unit %d", iterm->id); @@ -650,22 +650,22 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ while ((p1 = find_audio_control_unit(state, id)) != NULL) { term->id = id; switch (p1[2]) { - case INPUT_TERMINAL: + case UAC_INPUT_TERMINAL: term->type = combine_word(p1 + 4); term->channels = p1[7]; term->chconfig = combine_word(p1 + 8); term->name = p1[11]; return 0; - case FEATURE_UNIT: + case UAC_FEATURE_UNIT: id = p1[4]; break; /* continue to parse */ - case MIXER_UNIT: + case UAC_MIXER_UNIT: term->type = p1[2] << 16; /* virtual type */ term->channels = p1[5 + p1[4]]; term->chconfig = combine_word(p1 + 6 + p1[4]); term->name = p1[p1[0] - 1]; return 0; - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: /* call recursively to retrieve the channel info */ if (check_input_term(state, p1[5], term) < 0) return -ENODEV; @@ -673,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->id = id; term->name = p1[9 + p1[0] - 1]; return 0; - case PROCESSING_UNIT_V1: - case EXTENSION_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: if (p1[6] == 1) { id = p1[7]; break; /* continue to parse */ @@ -752,23 +752,23 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) break; } } - if (get_ctl_value(cval, GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || - get_ctl_value(cval, GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { + if (get_ctl_value(cval, UAC_GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || + get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n", cval->id, cval->mixer->ctrlif, cval->control, cval->id); return -EINVAL; } - if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { + if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { cval->res = 1; } else { int last_valid_res = cval->res; while (cval->res > 1) { - if (set_ctl_value(cval, SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) + if (set_ctl_value(cval, UAC_SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) break; cval->res /= 2; } - if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) + if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) cval->res = last_valid_res; } if (cval->res == 0) @@ -1097,7 +1097,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void struct uac_feature_unit_descriptor *ftr = _ftr; if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { - snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid); + snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } @@ -1739,17 +1739,17 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) } switch (p1[2]) { - case INPUT_TERMINAL: + case UAC_INPUT_TERMINAL: return 0; /* NOP */ - case MIXER_UNIT: + case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: return parse_audio_selector_unit(state, unitid, p1); - case FEATURE_UNIT: + case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case PROCESSING_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: return parse_audio_processing_unit(state, unitid, p1); - case EXTENSION_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); @@ -1779,7 +1779,7 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) /* * create mixer controls * - * walk through all OUTPUT_TERMINAL descriptors to search for mixers + * walk through all UAC_OUTPUT_TERMINAL descriptors to search for mixers */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { @@ -1807,7 +1807,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } desc = NULL; - while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) { + while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, UAC_OUTPUT_TERMINAL)) != NULL) { if (desc->bLength < 9) continue; /* invalid descriptor? */ set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ @@ -2047,7 +2047,7 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) } mixer->rc_setup_packet->bRequestType = USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE; - mixer->rc_setup_packet->bRequest = GET_MEM; + mixer->rc_setup_packet->bRequest = UAC_GET_MEM; mixer->rc_setup_packet->wValue = cpu_to_le16(0); mixer->rc_setup_packet->wIndex = cpu_to_le16(0); mixer->rc_setup_packet->wLength = cpu_to_le16(len); @@ -2170,7 +2170,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%s: ", jacks[i].name); err = snd_usb_ctl_msg(mixer->chip->dev, usb_rcvctrlpipe(mixer->chip->dev, 0), - GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | + UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE, 0, jacks[i].unitid << 8, buf, 3, 100); if (err == 3 && (buf[0] == 3 || buf[0] == 6)) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index fc1d2cd..f06faf7 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -91,7 +91,7 @@ .idVendor = 0x046d, .idProduct = 0x0850, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -100,7 +100,7 @@ .idVendor = 0x046d, .idProduct = 0x08ae, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -109,7 +109,7 @@ .idVendor = 0x046d, .idProduct = 0x08c6, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -118,7 +118,7 @@ .idVendor = 0x046d, .idProduct = 0x08f0, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -127,7 +127,7 @@ .idVendor = 0x046d, .idProduct = 0x08f5, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -136,7 +136,7 @@ .idVendor = 0x046d, .idProduct = 0x08f6, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { USB_DEVICE(0x046d, 0x0990), @@ -301,7 +301,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = EP_CS_ATTR_FILL_MAX, + .attributes = UAC_EP_CS_ATTR_FILL_MAX, .endpoint = 0x81, .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -2108,7 +2108,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2122,7 +2122,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2136,7 +2136,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2150,7 +2150,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2164,7 +2164,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2178,7 +2178,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2192,7 +2192,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2206,7 +2206,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-850", @@ -2238,7 +2238,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = EP_CS_ATTR_SAMPLE_RATE, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x02, .ep_attr = 0x01, .maxpacksize = 0x130, @@ -2268,7 +2268,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .match_flags = USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_MIDI_STREAMING, + .bInterfaceSubClass = USB_SUBCLASS_MIDISTREAMING, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_MIDI_STANDARD_INTERFACE diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb296..44deb21 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -16,6 +16,8 @@ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#include +#include #include #include #include @@ -315,9 +317,9 @@ static int us122l_set_sample_rate(struct usb_device *dev, int rate) data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000); + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000); if (err < 0) snd_printk(KERN_ERR "%d: cannot set freq %d to ep 0x%x\n", dev->devnum, rate, ep); -- cgit v0.10.2 From 76e6f5a9efc919f9179163c66403451a789d47ab Mon Sep 17 00:00:00 2001 From: Reimundo Heluani Date: Tue, 23 Feb 2010 01:19:51 -0800 Subject: ALSA: add support for Macbook Air 2,1 internal speaker Add support for Macbook Air 2,1 (late 2008) internal speaker and headphones. Create a "mba21" model for snd-hda-intel. Signed-off-by: Reimundo Heluani Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c74ca39..5382872 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -209,6 +209,7 @@ enum { ALC882_ASUS_A7J, ALC882_ASUS_A7M, ALC885_MACPRO, + ALC885_MBA21, ALC885_MBP3, ALC885_MB5, ALC885_MACMINI3, @@ -6948,6 +6949,13 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = { { 8, alc882_sixstack_ch8_init }, }; + +/* Macbook Air 2,1 */ + +static struct hda_channel_mode alc885_mba21_ch_modes[1] = { + { 2, NULL }, +}; + /* * macbook pro ALC885 can switch LineIn to LineOut without losing Mic */ @@ -7220,6 +7228,15 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { { } /* end */ }; +/* Macbook Air 2,1 same control for HP and internal Speaker */ + +static struct snd_kcontrol_new alc885_mba21_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT), + { } +}; + + static struct snd_kcontrol_new alc885_mbp3_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7689,6 +7706,29 @@ static struct hda_verb alc885_macmini3_init_verbs[] = { { } }; + +static struct hda_verb alc885_mba21_init_verbs[] = { + /*Internal and HP Speaker Mixer*/ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /*Internal Speaker Pin (0x0c)*/ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: output 0 (0x0e) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)}, + /* Line in (is hp when jack connected)*/ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + { } + }; + + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7854,6 +7894,17 @@ static void alc885_imac24_setup(struct hda_codec *codec) #define alc885_mb5_setup alc885_imac24_setup #define alc885_macmini3_setup alc885_imac24_setup +/* Macbook Air 2,1 */ +static void alc885_mba21_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; +} + + + static void alc885_mbp3_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9017,6 +9068,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", [ALC885_MACMINI3] = "macmini3", + [ALC885_MBA21] = "mba21", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9252,6 +9304,18 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, }, + [ALC885_MBA21] = { + .mixers = { alc885_mba21_mixer }, + .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs }, + .num_dacs = 2, + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mba21_ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), + .input_mux = &alc882_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_mba21_setup, + .init_hook = alc_automute_amp, + }, [ALC885_MBP3] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mbp3_init_verbs, -- cgit v0.10.2 From 0d7d8bf3b896dd8752284e2a7639a03936085e5f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Feb 2010 10:55:46 +0100 Subject: ALSA: hda - Add missing description in HD-Audio-Models.txt Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 5efacf0..1d38b0d 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -125,6 +125,7 @@ ALC882/883/885/888/889 macpro MacPro support mb5 Macbook 5,1 macmini3 Macmini 3,1 + mba21 Macbook Air 2,1 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection imac91 iMac 9,1 -- cgit v0.10.2 From e17dd32f342d0e876f729b348614320b297cf6f3 Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Mon, 22 Feb 2010 15:09:19 -0600 Subject: ASoC: OMAP: data_type and sync_mode configurable in audio dma Allow client drivers to set the data_type (16, 32) and the sync_mode (element, packet, etc) of the audio dma transferences. McBSP dai driver configures it for a data type of 16 bits and element sync mode. Signed-off-by: Misael Lopez Cruz Signed-off-by: Jorge Eduardo Candelaria Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6bbbd2a..d297256 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -287,6 +287,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; if (mcbsp_data->configured) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9db2770..825db38 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, @@ -149,6 +150,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) struct omap_runtime_data *prtd = runtime->private_data; struct omap_pcm_dma_data *dma_data = prtd->dma_data; struct omap_dma_channel_params dma_params; + int bytes; /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -156,11 +158,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) return 0; memset(&dma_params, 0, sizeof(dma_params)); - /* - * Note: Regardless of interface data formats supported by OMAP McBSP - * or EAC blocks, internal representation is always fixed 16-bit/sample - */ - dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; + dma_params.data_type = dma_data->data_type; dma_params.trigger = dma_data->dma_req; dma_params.sync_mode = dma_data->sync_mode; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -170,6 +168,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; dma_params.dst_port = OMAP_DMA_PORT_MPUI; + dma_params.dst_fi = dma_data->packet_size; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; @@ -177,6 +176,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; dma_params.src_port = OMAP_DMA_PORT_MPUI; + dma_params.src_fi = dma_data->packet_size; } /* * Set DMA transfer frame size equal to ALSA period size and frame @@ -184,7 +184,8 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) * we can transfer the whole ALSA buffer with single DMA transfer but * still can get an interrupt at each period bounary */ - dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2; + bytes = snd_pcm_lib_period_bytes(substream); + dma_params.elem_count = bytes >> dma_data->data_type; dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index 38a821d..b19975d 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -29,8 +29,10 @@ struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ unsigned long port_addr; /* transmit/receive register */ - int sync_mode; /* DMA sync mode */ void (*set_threshold)(struct snd_pcm_substream *substream); + int data_type; /* data type 8,16,32 */ + int sync_mode; /* DMA sync mode */ + int packet_size; /* packet size only in PACKET mode */ }; extern struct snd_soc_platform omap_soc_platform; -- cgit v0.10.2 From b3b0b4580bcb771d1d53b3d5acf689cba9907392 Mon Sep 17 00:00:00 2001 From: "Candelaria Villareal, Jorge" Date: Mon, 22 Feb 2010 17:17:21 -0600 Subject: ASoC: OMAP4: Add support for McPDM McPDM is the interface between Phoenix audio codec and the OMAP4430 processor. It enables data to be transfered to/from Phoenix at sample rates of 88.4 or 96 KHz. Signed-off-by: Jorge Eduardo Candelaria Signed-off-by: Margarita Olaya Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c new file mode 100644 index 0000000..ad8df6c --- /dev/null +++ b/sound/soc/omap/mcpdm.c @@ -0,0 +1,484 @@ +/* + * mcpdm.c -- McPDM interface driver + * + * Author: Jorge Eduardo Candelaria + * Copyright (C) 2009 - Texas Instruments, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "mcpdm.h" + +static struct omap_mcpdm *mcpdm; + +static inline void omap_mcpdm_write(u16 reg, u32 val) +{ + __raw_writel(val, mcpdm->io_base + reg); +} + +static inline int omap_mcpdm_read(u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} + +static void omap_mcpdm_reg_dump(void) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); +} + +/* + * Takes the McPDM module in and out of reset state. + * Uplink and downlink can be reset individually. + */ +static void omap_mcpdm_reset_capture(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +static void omap_mcpdm_reset_playback(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +void omap_mcpdm_start(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +void omap_mcpdm_stop(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Configures McPDM uplink for audio recording. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + int ctrl; + + if (!uplink) + return -EINVAL; + + mcpdm->uplink = uplink; + + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; + + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Configures McPDM downlink for audio playback. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + int ctrl; + + if (!downlink) + return -EINVAL; + + mcpdm->downlink = downlink; + + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; + + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Cleans McPDM uplink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + + if (!uplink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + + /* Clear Downlink channels */ + mcpdm->up_channels = 0; + + mcpdm->uplink = NULL; + + return 0; +} + +/* + * Cleans McPDM downlink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + + if (!downlink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + + /* clear Downlink channels */ + mcpdm->dn_channels = 0; + + mcpdm->downlink = NULL; + + return 0; +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + if (irq & MCPDM_DN_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ) { + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + } + + if (irq & MCPDM_UP_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ) { + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + } + + return IRQ_HANDLED; +} + +int omap_mcpdm_request(void) +{ + int ret; + + clk_enable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + ret = -EBUSY; + goto err; + } + mcpdm->free = 0; + + spin_unlock(&mcpdm->lock); + + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + goto err; + } + + return 0; + +err: + clk_disable(mcpdm->clk); + return ret; +} + +void omap_mcpdm_free(void) +{ + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + clk_disable(mcpdm->clk); + + free_irq(mcpdm->irq, (void *)mcpdm); +} + +/* Enable/disable DC offset cancelation for the analog + * headset path (PDM channels 1 and 2). + */ +int omap_mcpdm_set_offset(int offset1, int offset2) +{ + int offset; + + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; + + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + + /* offset cancellation for channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; + + /* offset cancellation for channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; + + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + + return 0; +} + +static int __devinit omap_mcpdm_probe(struct platform_device *pdev) +{ + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_resource; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_resource; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + + mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); + if (IS_ERR(mcpdm->clk)) { + ret = PTR_ERR(mcpdm->clk); + dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); + goto err_clk; + } + + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; + +err_clk: + iounmap(mcpdm->io_base); +err_resource: + kfree(mcpdm); +exit: + return ret; +} + +static int __devexit omap_mcpdm_remove(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + + platform_set_drvdata(pdev, NULL); + + clk_put(mcpdm_ptr->clk); + + iounmap(mcpdm_ptr->io_base); + + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; + + kfree(mcpdm_ptr); + + return 0; +} + +static struct platform_driver omap_mcpdm_driver = { + .probe = omap_mcpdm_probe, + .remove = __devexit_p(omap_mcpdm_remove), + .driver = { + .name = "omap-mcpdm", + }, +}; + +static struct platform_device *omap_mcpdm_device; + +static int __init omap_mcpdm_init(void) +{ + return platform_driver_register(&omap_mcpdm_driver); +} +arch_initcall(omap_mcpdm_init); diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h new file mode 100644 index 0000000..7bb326e --- /dev/null +++ b/sound/soc/omap/mcpdm.h @@ -0,0 +1,151 @@ +/* + * mcpdm.h -- Defines for McPDM driver + * + * Author: Jorge Eduardo Candelaria + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +/* McPDM registers */ + +#define MCPDM_REVISION 0x00 +#define MCPDM_SYSCONFIG 0x10 +#define MCPDM_IRQSTATUS_RAW 0x24 +#define MCPDM_IRQSTATUS 0x28 +#define MCPDM_IRQENABLE_SET 0x2C +#define MCPDM_IRQENABLE_CLR 0x30 +#define MCPDM_IRQWAKE_EN 0x34 +#define MCPDM_DMAENABLE_SET 0x38 +#define MCPDM_DMAENABLE_CLR 0x3C +#define MCPDM_DMAWAKEEN 0x40 +#define MCPDM_CTRL 0x44 +#define MCPDM_DN_DATA 0x48 +#define MCPDM_UP_DATA 0x4C +#define MCPDM_FIFO_CTRL_DN 0x50 +#define MCPDM_FIFO_CTRL_UP 0x54 +#define MCPDM_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define DMA_DN_ENABLE 0x1 +#define DMA_UP_ENABLE 0x2 + +/* + * MCPDM_CTRL bit fields + */ + +#define PDM_UP1_EN 0x0001 +#define PDM_UP2_EN 0x0002 +#define PDM_UP3_EN 0x0004 +#define PDM_DN1_EN 0x0008 +#define PDM_DN2_EN 0x0010 +#define PDM_DN3_EN 0x0020 +#define PDM_DN4_EN 0x0040 +#define PDM_DN5_EN 0x0080 +#define PDMOUTFORMAT 0x0100 +#define CMD_INT 0x0200 +#define STATUS_INT 0x0400 +#define SW_UP_RST 0x0800 +#define SW_DN_RST 0x1000 +#define PDM_UP_MASK 0x007 +#define PDM_DN_MASK 0x0F8 +#define PDM_CMD_MASK 0x200 +#define PDM_STATUS_MASK 0x400 + + +#define PDMOUTFORMAT_LJUST (0 << 8) +#define PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define UP_THRES_MAX 0xF +#define DN_THRES_MAX 0xF + +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define DN_OFST_RX1_EN 0x0001 +#define DN_OFST_RX2_EN 0x0100 + +#define DN_OFST_RX1 1 +#define DN_OFST_RX2 9 +#define DN_OFST_MAX 0x1F + +#define MCPDM_UPLINK 1 +#define MCPDM_DOWNLINK 2 + +struct omap_mcpdm_link { + int irq_mask; + int threshold; + int format; + int channels; +}; + +struct omap_mcpdm_platform_data { + unsigned long phys_base; + u16 irq; +}; + +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + u8 free; + int irq; + + spinlock_t lock; + struct omap_mcpdm_platform_data *pdata; + struct clk *clk; + struct omap_mcpdm_link *downlink; + struct omap_mcpdm_link *uplink; + struct completion irq_completion; + + int dn_channels; + int up_channels; +}; + +extern void omap_mcpdm_start(int stream); +extern void omap_mcpdm_stop(int stream); +extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_request(void); +extern void omap_mcpdm_free(void); +extern int omap_mcpdm_set_offset(int offset1, int offset2); -- cgit v0.10.2 From db72c2f89790f919d65d0adbee390958005c40fc Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Mon, 22 Feb 2010 15:09:22 -0600 Subject: ASoC: OMAP4: Add McPDM platform driver McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz Signed-off-by: Margarita Olaya Signed-off-by: Jorge Eduardo Candelaria Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 18ebdc7..f11963c 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP +config SND_OMAP_SOC_MCPDM + tristate + config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 19283e5..0bc00ca 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,9 +1,11 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 0000000..25f19e4 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,251 @@ +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 Texas Instruments + * + * Author: Misael Lopez Cruz + * Contact: Jorge Eduardo Candelaria + * Margarita Olaya + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm_data { + struct omap_mcpdm_link *links; + int active; +}; + +static struct omap_mcpdm_link omap_mcpdm_links[] = { + /* downlink */ + { + .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, +}; + +static struct omap_mcpdm_data mcpdm_data = { + .links = omap_mcpdm_links, + .active = 0, +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio playback", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio capture", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA, + }, +}; + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err = 0; + + if (!cpu_dai->active) + err = omap_mcpdm_request(); + + return err; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (!cpu_dai->active) + omap_mcpdm_free(); +} + +static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + int stream = substream->stream; + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcpdm_priv->active++) + omap_mcpdm_start(stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcpdm_priv->active) + omap_mcpdm_stop(stream); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int channels, err, link_mask = 0; + + cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_playback_open(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_capture_open(&mcpdm_links[stream]); + } + + return err; +} + +static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int err; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + err = omap_mcpdm_playback_close(&mcpdm_links[stream]); + else + err = omap_mcpdm_capture_close(&mcpdm_links[stream]); + + return err; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .trigger = omap_mcpdm_dai_trigger, + .hw_params = omap_mcpdm_dai_hw_params, + .hw_free = omap_mcpdm_dai_hw_free, +}; + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai omap_mcpdm_dai = { + .name = "omap-mcpdm", + .id = -1, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, + .private_data = &mcpdm_data, +}; +EXPORT_SYMBOL_GPL(omap_mcpdm_dai); + +static int __init snd_omap_mcpdm_init(void) +{ + return snd_soc_register_dai(&omap_mcpdm_dai); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + snd_soc_unregister_dai(&omap_mcpdm_dai); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz "); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..73b80d5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,29 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 Texas Instruments + * + * Contact: Misael Lopez Cruz + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +extern struct snd_soc_dai omap_mcpdm_dai; + +#endif /* End of __OMAP_MCPDM_H__ */ -- cgit v0.10.2 From 47fc9a0a808f23b7b305f6c018e4882118b88d92 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 22 Feb 2010 16:41:57 +0900 Subject: ASoC: fsi: Modify over/under run error settlement In current FSI driver, playback function cares only overrun, and capture function cares only underrun. But playback function should had cared about underrun, and capture function should had cared about overrun too. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c36d24..993abb7 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -388,7 +388,7 @@ static void fsi_soft_all_reset(struct fsi_master *master) } /* playback interrupt */ -static int fsi_data_push(struct fsi_priv *fsi) +static int fsi_data_push(struct fsi_priv *fsi, int startup) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -397,7 +397,7 @@ static int fsi_data_push(struct fsi_priv *fsi) int fifo_free; int width; u8 *start; - int i, ret, over_period; + int i, over_period; if (!fsi || !fsi->substream || @@ -453,24 +453,26 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi->byte_offset += send * width; - ret = 0; status = fsi_reg_read(fsi, DOFF_ST); - if (status & ERR_OVER) { + if (!startup) { struct snd_soc_dai *dai = fsi_get_dai(substream); - dev_err(dai->dev, "over run error\n"); - fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR); - ret = -EIO; + + if (status & ERR_OVER) + dev_err(dai->dev, "over run\n"); + if (status & ERR_UNDER) + dev_err(dai->dev, "under run\n"); } + fsi_reg_write(fsi, DOFF_ST, 0); fsi_irq_enable(fsi, 1); if (over_period) snd_pcm_period_elapsed(substream); - return ret; + return 0; } -static int fsi_data_pop(struct fsi_priv *fsi) +static int fsi_data_pop(struct fsi_priv *fsi, int startup) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -479,7 +481,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) int fifo_fill; int width; u8 *start; - int i, ret, over_period; + int i, over_period; if (!fsi || !fsi->substream || @@ -534,21 +536,23 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi->byte_offset += fifo_fill * width; - ret = 0; status = fsi_reg_read(fsi, DIFF_ST); - if (status & ERR_UNDER) { + if (!startup) { struct snd_soc_dai *dai = fsi_get_dai(substream); - dev_err(dai->dev, "under run error\n"); - fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR); - ret = -EIO; + + if (status & ERR_OVER) + dev_err(dai->dev, "over run\n"); + if (status & ERR_UNDER) + dev_err(dai->dev, "under run\n"); } + fsi_reg_write(fsi, DIFF_ST, 0); fsi_irq_enable(fsi, 0); if (over_period) snd_pcm_period_elapsed(substream); - return ret; + return 0; } static irqreturn_t fsi_interrupt(int irq, void *data) @@ -562,13 +566,13 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_master_write(master, SOFT_RST, status | 0x00000010); if (int_st & INT_A_OUT) - fsi_data_push(&master->fsia); + fsi_data_push(&master->fsia, 0); if (int_st & INT_B_OUT) - fsi_data_push(&master->fsib); + fsi_data_push(&master->fsib, 0); if (int_st & INT_A_IN) - fsi_data_pop(&master->fsia); + fsi_data_pop(&master->fsia, 0); if (int_st & INT_B_IN) - fsi_data_pop(&master->fsib); + fsi_data_pop(&master->fsib, 0); fsi_master_write(master, INT_ST, 0x0000000); @@ -726,7 +730,7 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, fsi_stream_push(fsi, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); + ret = is_play ? fsi_data_push(fsi, 1) : fsi_data_pop(fsi, 1); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); -- cgit v0.10.2 From aefbd3e823d4fe219bb6420b0cac505847270507 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 23 Feb 2010 10:30:00 +0100 Subject: usb/gadget/{f_audio,gmidi}.c: follow recent changes in audio.h Some structs in linux/usb/audio.h have got new names to mark them as part of version 1.0 of the USB audio standard. Follow these changes in the gadget drivers. Note that this header and the ALSA USB driver will undergo some refactoring soon, so there might be another update to the gadgets as well. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/drivers/usb/gadget/f_audio.c b/drivers/usb/gadget/f_audio.c index df77f61..f1e3aad 100644 --- a/drivers/usb/gadget/f_audio.c +++ b/drivers/usb/gadget/f_audio.c @@ -60,7 +60,7 @@ DECLARE_UAC_AC_HEADER_DESCRIPTOR(2); #define UAC_DT_TOTAL_LENGTH (UAC_DT_AC_HEADER_LENGTH + UAC_DT_INPUT_TERMINAL_SIZE \ + UAC_DT_OUTPUT_TERMINAL_SIZE + UAC_DT_FEATURE_UNIT_SIZE(0)) /* B.3.2 Class-Specific AC Interface Descriptor */ -static struct uac_ac_header_descriptor_2 ac_header_desc = { +static struct uac_ac_header_descriptor_v1_2 ac_header_desc = { .bLength = UAC_DT_AC_HEADER_LENGTH, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_HEADER, @@ -124,7 +124,7 @@ static struct usb_audio_control_selector feature_unit = { }; #define OUTPUT_TERMINAL_ID 3 -static struct uac_output_terminal_descriptor output_terminal_desc = { +static struct uac_output_terminal_descriptor_v1 output_terminal_desc = { .bLength = UAC_DT_OUTPUT_TERMINAL_SIZE, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_OUTPUT_TERMINAL, @@ -154,7 +154,7 @@ static struct usb_interface_descriptor as_interface_alt_1_desc = { }; /* B.4.2 Class-Specific AS Interface Descriptor */ -static struct uac_as_header_descriptor as_header_desc = { +static struct uac_as_header_descriptor_v1 as_header_desc = { .bLength = UAC_DT_AS_HEADER_SIZE, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_AS_GENERAL, diff --git a/drivers/usb/gadget/gmidi.c b/drivers/usb/gadget/gmidi.c index d0b1e83..5f6a2e0 100644 --- a/drivers/usb/gadget/gmidi.c +++ b/drivers/usb/gadget/gmidi.c @@ -237,7 +237,7 @@ static const struct usb_interface_descriptor ac_interface_desc = { }; /* B.3.2 Class-Specific AC Interface Descriptor */ -static const struct uac_ac_header_descriptor_1 ac_header_desc = { +static const struct uac_ac_header_descriptor_v1_1 ac_header_desc = { .bLength = UAC_DT_AC_HEADER_SIZE(1), .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = USB_MS_HEADER, -- cgit v0.10.2 From dd2b4a7abf82d88261f8f98e1361388a7db2ffe4 Mon Sep 17 00:00:00 2001 From: "Zhang, Rui" Date: Wed, 24 Feb 2010 09:38:49 +0800 Subject: ALSA: hda - remove unnecessary msleep on power state transitions This will save ~15ms boot time. The first 10ms sleep was introduced in commit d2595d86e5 for (buggy) Cxt codecs, so better to limit the sleep to the problem hardware. For the second 10ms sleep, the HDA spec says: Power State[1:0]: 00: Node Power state (D0) is fully on. 01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog playback) which must remain fully on. 10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state. 11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software control. Note that any low power state set by software must retain sufficient operational capability to properly respond to subsequent software Power State command. So 10ms is actually the max wait time. It should be safe to remove/reduce it and rely on the loop of 1ms-sleeps. CC: Marc Boucher CC: Arjan van de Ven Signed-off-by: Zhang Rui Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 98767df..76d3c4c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2767,7 +2767,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); /* partial workaround for "azx_get_response timeout" */ - if (power_state == AC_PWRST_D0) + if (power_state == AC_PWRST_D0 && + (codec->vendor_id & 0xffff0000) == 0x14f10000) msleep(10); nid = codec->start_nid; @@ -2801,7 +2802,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (power_state == AC_PWRST_D0) { unsigned long end_time; int state; - msleep(10); /* wait until the codec reachs to D0 */ end_time = jiffies + msecs_to_jiffies(500); do { -- cgit v0.10.2 From d62abe563fa4718e7f85f3e871655434db92366d Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Tue, 23 Feb 2010 18:10:19 -0600 Subject: OMAP4: PMIC: Add support for twl6030 codec In order to have TWL6030 CODEC driver as a platform driver, codec data should be passed through twl_platform_data structure. For twl6030 audio codec, the following data may be passed: - audpwron_gpio: gpio line used to power-up/down the codec. A low-to-high transition powers codec up. Setting audpwron_gpio to a negative value means that codec will use manual power sequence instead of automatic sequence - naudint_irq: irq line for audio interrupt. twl6030 drives NAUDINT line to low when an interrupt (codec ready, plug insertion/removal, etc) is detected However, codec driver can operate if any or none of them are passed. Signed-off-by: Misael Lopez Cruz Signed-off-by: Margarita Olaya Cabrera Signed-off-by: Jorge Eduardo Candelaria Acked-by: Samuel Ortiz Acked-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index 2a76065..19a930d 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -115,7 +115,8 @@ #define twl_has_watchdog() false #endif -#if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) +#if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\ + defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) #define twl_has_codec() true #else #define twl_has_codec() false @@ -711,8 +712,19 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features) return PTR_ERR(child); } - if (twl_has_codec() && pdata->codec) { - child = add_child(1, "twl4030_codec", + if (twl_has_codec() && pdata->codec && twl_class_is_4030()) { + sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid; + child = add_child(sub_chip_id, "twl4030_codec", + pdata->codec, sizeof(*pdata->codec), + false, 0, 0); + if (IS_ERR(child)) + return PTR_ERR(child); + } + + /* Phoenix*/ + if (twl_has_codec() && pdata->codec && twl_class_is_6030()) { + sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid; + child = add_child(sub_chip_id, "twl6030_codec", pdata->codec, sizeof(*pdata->codec), false, 0, 0); if (IS_ERR(child)) diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index bf1c5be..7897f30 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -547,6 +547,10 @@ struct twl4030_codec_data { unsigned int audio_mclk; struct twl4030_codec_audio_data *audio; struct twl4030_codec_vibra_data *vibra; + + /* twl6030 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ }; struct twl4030_platform_data { -- cgit v0.10.2 From 6227cdced0328b0c4322c3170a727af5249393ce Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 25 Feb 2010 08:36:52 +0100 Subject: ALSA: hda - Add ALC670 codec support - Fixed alc_subsystem_id( ) typo and add new function. - !(ass & 0x100000)) ==> Delete this check. It is unnecessary check. - Add porti - ALC670 support Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5382872..220a49f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1254,7 +1254,7 @@ static void alc_init_auto_mic(struct hda_codec *codec) */ static int alc_subsystem_id(struct hda_codec *codec, hda_nid_t porta, hda_nid_t porte, - hda_nid_t portd) + hda_nid_t portd, hda_nid_t porti) { unsigned int ass, tmp, i; unsigned nid; @@ -1280,7 +1280,7 @@ static int alc_subsystem_id(struct hda_codec *codec, snd_printd("realtek: No valid SSID, " "checking pincfg 0x%08x for NID 0x%x\n", ass, nid); - if (!(ass & 1) && !(ass & 0x100000)) + if (!(ass & 1)) return 0; if ((ass >> 30) != 1) /* no physical connection */ return 0; @@ -1340,6 +1340,8 @@ do_sku: nid = porte; else if (tmp == 2) nid = portd; + else if (tmp == 3) + nid = porti; else return 1; for (i = 0; i < spec->autocfg.line_outs; i++) @@ -1354,9 +1356,10 @@ do_sku: } static void alc_ssid_check(struct hda_codec *codec, - hda_nid_t porta, hda_nid_t porte, hda_nid_t portd) + hda_nid_t porta, hda_nid_t porte, + hda_nid_t portd, hda_nid_t porti) { - if (!alc_subsystem_id(codec, porta, porte, portd)) { + if (!alc_subsystem_id(codec, porta, porte, portd, porti)) { struct alc_spec *spec = codec->spec; snd_printd("realtek: " "Enable default setup for auto mode as fallback\n"); @@ -4859,7 +4862,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -6393,7 +6396,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - alc_ssid_check(codec, 0x10, 0x15, 0x0f); + alc_ssid_check(codec, 0x10, 0x15, 0x0f, 0); return 1; } @@ -10224,7 +10227,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -11782,7 +11785,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - alc_ssid_check(codec, 0x15, 0x14, 0x1b); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -12733,7 +12736,6 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: - case 0x21: dac = 0x03; break; default: @@ -12954,7 +12956,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -13845,11 +13847,11 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { add_verb(spec, alc269vb_init_verbs); real_capsrc_nids = alc269vb_capsrc_nids[0]; - alc_ssid_check(codec, 0x21, 0x1b, 0x14); + alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21); } else { add_verb(spec, alc269_init_verbs); real_capsrc_nids = alc269_capsrc_nids[0]; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); } spec->num_mux_defs = 1; @@ -15013,7 +15015,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); set_capture_mixer(codec); - alc_ssid_check(codec, 0x0e, 0x0f, 0x0b); + alc_ssid_check(codec, 0x0e, 0x0f, 0x0b, 0); return 1; } @@ -15904,7 +15906,7 @@ static struct alc_config_preset alc861vd_presets[] = { static int alc861vd_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0); + return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0); } @@ -16140,7 +16142,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -17627,6 +17629,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), @@ -18257,7 +18260,11 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - alc_ssid_check(codec, 0x15, 0x1b, 0x14); + if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0x21); + else + alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); return 1; } @@ -18407,6 +18414,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, + { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, -- cgit v0.10.2 From 61c2d2b5e7241d4410ab8227ef4f76c1aba8210b Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 25 Feb 2010 08:49:06 +0100 Subject: ALSA: hda - Add/fix ALC269 FSC and Quanta models Specify proper quirk models for FSC and Quanta machines with ALC269 codec. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 220a49f..e8cbe21 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13946,8 +13946,14 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), - SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), + SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), + SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), + SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), + SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC), + SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC), + SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC), {} }; -- cgit v0.10.2 From 9e4a10d27e89f780539e08abd2b051cb83635dfa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 25 Feb 2010 12:52:09 +0000 Subject: ASoC: Remove a unused variables from i.MX FIQ runtime data Signed-off-by: Mark Brown Acked-by: Sascha Hauer Acked-by: Liam Girdwood diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 5532579..a1c4ce6 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -35,12 +35,8 @@ struct imx_pcm_runtime_data { int period; int periods; - unsigned long dma_addr; - int dma; unsigned long offset; unsigned long size; - unsigned long period_cnt; - void *buf; struct timer_list timer; int period_time; }; -- cgit v0.10.2 From b4e82b5b785670b68136765059d1afc65c0ae023 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 25 Feb 2010 12:52:10 +0000 Subject: ASoC: Check progress when reporting periods from i.MX FIQ handler Currently the i.MX FIQ handler is reporting periods as elapsed based purely on a timer running in the CPU. This means that any clock mismatch between the CPU and the audio subsystem can result in the status reported to applications drifting away from the actual status of the hardware. This is particularly likely at present since the SSI driver is only capable of operating in slave mode so it's very likely that the interface will be clocked from a different source. Instead check the offset reported by the FIQ and only notify when we have transferred at least one period, re-firing the timer if we didn't do so. Also factor out the calculation of the timer expiry time for make it a bit easier to experiment with. Note that this only improves the situation, problems can still be triggered. Signed-off-by: Mark Brown Acked-by: Sascha Hauer Acked-by: Liam Girdwood diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index a1c4ce6..d9cb984 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -36,17 +36,24 @@ struct imx_pcm_runtime_data { int period; int periods; unsigned long offset; + unsigned long last_offset; unsigned long size; struct timer_list timer; - int period_time; + int poll_time; }; +static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd) +{ + iprtd->timer.expires = jiffies + iprtd->poll_time; +} + static void imx_ssi_timer_callback(unsigned long data) { struct snd_pcm_substream *substream = (void *)data; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; struct pt_regs regs; + unsigned long delta; get_fiq_regs(®s); @@ -55,9 +62,25 @@ static void imx_ssi_timer_callback(unsigned long data) else iprtd->offset = regs.ARM_r9 & 0xffff; - iprtd->timer.expires = jiffies + iprtd->period_time; + /* How much data have we transferred since the last period report? */ + if (iprtd->offset >= iprtd->last_offset) + delta = iprtd->offset - iprtd->last_offset; + else + delta = runtime->buffer_size + iprtd->offset + - iprtd->last_offset; + + /* If we've transferred at least a period then report it and + * reset our poll time */ + if (delta >= runtime->period_size) { + snd_pcm_period_elapsed(substream); + iprtd->last_offset = iprtd->offset; + + imx_ssi_set_next_poll(iprtd); + } + + /* Restart the timer; if we didn't report we'll run on the next tick */ add_timer(&iprtd->timer); - snd_pcm_period_elapsed(substream); + } static struct fiq_handler fh = { @@ -72,9 +95,10 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params); + iprtd->period = params_period_bytes(params) ; iprtd->offset = 0; - iprtd->period_time = HZ / (params_rate(params) / params_period_size(params)); + iprtd->last_offset = 0; + iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params)); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); @@ -110,7 +134,7 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - iprtd->timer.expires = jiffies + iprtd->period_time; + imx_ssi_set_next_poll(iprtd); add_timer(&iprtd->timer); if (++fiq_enable == 1) enable_fiq(imx_pcm_fiq); -- cgit v0.10.2