From 393a53cbcfab404d516544cd30650c7b104cd3d4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 4 Oct 2012 15:08:38 +0200 Subject: ASoC: enable 192KHz support for McASP Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 0de9ed6..156f15f 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -23,7 +23,7 @@ #include "davinci-pcm.h" -#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000 +#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 #define DAVINCI_MCASP_I2S_DAI 0 #define DAVINCI_MCASP_DIT_DAI 1 diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 93ea3bf..6ebbc88 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -67,13 +67,9 @@ static struct snd_pcm_hardware pcm_hardware_playback = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| SNDRV_PCM_INFO_BATCH), .formats = DAVINCI_PCM_FMTBITS, - .rates = (SNDRV_PCM_RATE_8000 | 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 | - SNDRV_PCM_RATE_KNOT), + .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT, .rate_min = 8000, - .rate_max = 96000, + .rate_max = 192000, .channels_min = 2, .channels_max = 384, .buffer_bytes_max = 128 * 1024, @@ -90,13 +86,9 @@ static struct snd_pcm_hardware pcm_hardware_capture = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_BATCH), .formats = DAVINCI_PCM_FMTBITS, - .rates = (SNDRV_PCM_RATE_8000 | 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 | - SNDRV_PCM_RATE_KNOT), + .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT, .rate_min = 8000, - .rate_max = 96000, + .rate_max = 192000, .channels_min = 2, .channels_max = 384, .buffer_bytes_max = 128 * 1024, -- cgit v0.10.2 From 4ed8c9b737b61030d7b6ac71294d698de85b5b7e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 4 Oct 2012 15:08:39 +0200 Subject: ASoC: McASP: add support for clock dividers Add support for the internal clock dividers of the McASP driver. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 714e51e..9b1920e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -199,6 +199,7 @@ #define ACLKXE BIT(5) #define TX_ASYNC BIT(6) #define ACLKXPOL BIT(7) +#define ACLKXDIV_MASK 0x1f /* * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits @@ -207,6 +208,7 @@ #define ACLKRE BIT(5) #define RX_ASYNC BIT(6) #define ACLKRPOL BIT(7) +#define ACLKRDIV_MASK 0x1f /* * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control @@ -215,6 +217,7 @@ #define AHCLKXDIV(val) (val) #define AHCLKXPOL BIT(14) #define AHCLKXE BIT(15) +#define AHCLKXDIV_MASK 0xfff /* * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control @@ -223,6 +226,7 @@ #define AHCLKRDIV(val) (val) #define AHCLKRPOL BIT(14) #define AHCLKRE BIT(15) +#define AHCLKRDIV_MASK 0xfff /* * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits @@ -554,6 +558,32 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + + switch (div_id) { + case 0: /* MCLK divider */ + mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXDIV(div - 1), AHCLKXDIV_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRDIV(div - 1), AHCLKRDIV_MASK); + break; + + case 1: /* BCLK divider */ + mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXDIV(div - 1), ACLKXDIV_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKRCTL_REG, + ACLKRDIV(div - 1), ACLKRDIV_MASK); + break; + + default: + return -EINVAL; + } + + return 0; +} + static int davinci_config_channel_size(struct davinci_audio_dev *dev, int channel_size) { @@ -880,7 +910,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .trigger = davinci_mcasp_trigger, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, - + .set_clkdiv = davinci_mcasp_set_clkdiv, }; #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ -- cgit v0.10.2 From 5b66aa2d0cafd8093e0b8959d32e694c92390b8c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 4 Oct 2012 15:08:41 +0200 Subject: ASoC: McASP: make AHCLK direction configurable Add a .set_sysclk function to pass the direction of the clock down to the driver. Only enable AHCLKX in the PDIR register when the CPU is driving the clock. This also removes the modification of the AHCLKXE/AHCLKRE bits in the hw_params callback, and users must set the desired configuration using snd_soc_dai_set_sysclk(), which this patch also does for the only user in mainline (davinci-evm). Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 6fac5af..d55e647 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -71,6 +71,11 @@ static int evm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set the CPU system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9b1920e..9f7c2e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -486,8 +486,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, ACLKX | AFSX); break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ @@ -584,6 +583,24 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div return 0; } +static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_OUT) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + } else { + mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX); + } + + return 0; +} + static int davinci_config_channel_size(struct davinci_audio_dev *dev, int channel_size) { @@ -739,8 +756,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, - AHCLKXE); mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); @@ -756,8 +771,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* bit stream is MSB first with no delay */ /* DSP_B mode */ mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); - mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, - AHCLKRE); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) @@ -911,6 +924,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, .set_clkdiv = davinci_mcasp_set_clkdiv, + .set_sysclk = davinci_mcasp_set_sysclk, }; #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ -- cgit v0.10.2 From 5296cf2d5e245ca3373207ba3f0aedaa10f040b6 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 4 Oct 2012 15:08:42 +0200 Subject: ASoC: McASP: set format parameters in dependence of the DAI format The FSDUR flag configures whether the frame clock uses a high phase of only one bit or a full word. This has to be set depending on the DAI format. For other modes than DSP_B, the FSXDLY/FSRDLY fields have to be set to 1. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9f7c2e1..11728cd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -477,6 +477,23 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); void __iomem *base = dev->base; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_AC97: + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + break; + default: + /* configure a full-word SYNC pulse (LRCLK) */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + + /* make 1st data bit occur one ACLK cycle after the frame sync */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1)); + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1)); + break; + } + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* codec is clock and frame slave */ @@ -765,8 +782,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) else printk(KERN_ERR "playback tdm slot %d not supported\n", dev->tdm_slots); - - mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); } else { /* bit stream is MSB first with no delay */ /* DSP_B mode */ @@ -779,8 +794,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) else printk(KERN_ERR "capture tdm slot %d not supported\n", dev->tdm_slots); - - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); } } -- cgit v0.10.2 From 21eb24d8d50a26fe6bd626bc9771a469e14907dc Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 9 Oct 2012 09:35:16 +0200 Subject: ASoC: McASP: add support for 24 bit samples Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 11728cd..05a2b1e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -865,6 +865,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, word_length = DAVINCI_AUDIO_WORD_16; break; + case SNDRV_PCM_FORMAT_U24_3LE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: + dma_params->data_type = 3; + word_length = DAVINCI_AUDIO_WORD_24; + break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; @@ -944,6 +952,10 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE) -- cgit v0.10.2 From 6b7fa011571d10309cd2b39511f895cef9dd536c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 9 Oct 2012 11:56:40 +0200 Subject: ASoC: McASP: treat SNDRV_PCM_FORMAT_U24_* as 32 bit words Signed-off-by: Daniel Mack Reported-by: Mike Looijmans Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 05a2b1e..4fd149b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -867,12 +867,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: - case SNDRV_PCM_FORMAT_U24_LE: - case SNDRV_PCM_FORMAT_S24_LE: dma_params->data_type = 3; word_length = DAVINCI_AUDIO_WORD_24; break; + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; -- cgit v0.10.2 From b8ec56d8eabc6382bd6586024138d9f1d1121ceb Mon Sep 17 00:00:00 2001 From: Matt Porter Date: Wed, 17 Oct 2012 16:08:03 +0200 Subject: ASoC: davinci: replace private sram api with genalloc Removes the DaVinci private SRAM API and replaces it with the genalloc API. The SRAM gen_pool is passed in pdata since DaVinci is in the early stages of DT conversion. [zonque@gmail.com: stub out gen_pool functions for !CONFIG_GENERIC_ALLOCATOR] Signed-off-by: Matt Porter Signed-off-by: Mark Brown diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index d0c5825..f3d6e4f 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -16,6 +16,8 @@ #ifndef __DAVINCI_ASP_H #define __DAVINCI_ASP_H +#include + struct snd_platform_data { u32 tx_dma_offset; u32 rx_dma_offset; @@ -30,6 +32,7 @@ struct snd_platform_data { unsigned enable_channel_combine:1; unsigned sram_size_playback; unsigned sram_size_capture; + struct gen_pool *sram_pool; /* * If McBSP peripheral gets the clock from an external pin, diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4fd149b..5715595 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1167,6 +1167,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_playback; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + mem->start); @@ -1184,6 +1185,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_capture; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + mem->start); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 6ebbc88..afab81f 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include @@ -23,7 +24,6 @@ #include #include -#include #include "davinci-pcm.h" @@ -251,7 +251,9 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) } } -static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, +#ifdef CONFIG_GENERIC_ALLOCATOR +static int allocate_sram(struct snd_pcm_substream *substream, + struct gen_pool *sram_pool, unsigned size, struct snd_pcm_hardware *ppcm) { struct snd_dma_buffer *buf = &substream->dma_buffer; @@ -263,9 +265,10 @@ static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, return 0; ppcm->period_bytes_max = size; - iram_virt = sram_alloc(size, &iram_phys); + iram_virt = (void *)gen_pool_alloc(sram_pool, size); if (!iram_virt) goto exit1; + iram_phys = gen_pool_virt_to_phys(sram_pool, (unsigned)iram_virt); iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); if (!iram_dma) goto exit2; @@ -277,11 +280,33 @@ static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, return 0; exit2: if (iram_virt) - sram_free(iram_virt, size); + gen_pool_free(sram_pool, (unsigned)iram_virt, size); exit1: return -ENOMEM; } +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct gen_pool *sram_pool = prtd->params->sram_pool; + + gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); +} +#else +static int allocate_sram(struct snd_pcm_substream *substream, + struct gen_pool *sram_pool, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + return 0; +} + +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ +} +#endif + /* * Only used with ping/pong. * This is called after runtime->dma_addr, period_bytes and data_type are valid @@ -668,7 +693,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, params->sram_size, ppcm); + allocate_sram(substream, params->sram_pool, params->sram_size, ppcm); snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, @@ -811,7 +836,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) buf->area = NULL; iram_dma = buf->private_data; if (iram_dma) { - sram_free(iram_dma->area, iram_dma->bytes); + davinci_free_sram(substream, iram_dma); kfree(iram_dma); } } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fc4d01c..b6ef703 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,6 +12,7 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include #include #include @@ -20,6 +21,7 @@ struct davinci_pcm_dma_params { unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ unsigned sram_size; + struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */ enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned char data_type; /* xfer data type */ -- cgit v0.10.2 From 88abfd5b84cc98c839fb7783c59b4c79d63d45e2 Mon Sep 17 00:00:00 2001 From: Matt Porter Date: Wed, 17 Oct 2012 16:08:04 +0200 Subject: ARM: davinci: enable SRAM ping ping buffering on DA850 Passes the DA850 shared SRAM gen_pool to the McASP driver and enables the ping-pong buffer DMA support. Signed-off-by: Matt Porter Signed-off-by: Mark Brown diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index 32ee3f8..d9bc3fa 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -762,16 +762,19 @@ static u8 da850_iis_serializer_direction[] = { }; static struct snd_platform_data da850_evm_snd_data = { - .tx_dma_offset = 0x2000, - .rx_dma_offset = 0x2000, - .op_mode = DAVINCI_MCASP_IIS_MODE, - .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), - .tdm_slots = 2, - .serial_dir = da850_iis_serializer_direction, - .asp_chan_q = EVENTQ_0, - .version = MCASP_VERSION_2, - .txnumevt = 1, - .rxnumevt = 1, + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da850_iis_serializer_direction, + .asp_chan_q = EVENTQ_0, + .ram_chan_q = EVENTQ_1, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, + .sram_size_playback = SZ_8K, + .sram_size_capture = SZ_8K, }; static const short da850_evm_mcasp_pins[] __initconst = { @@ -1509,6 +1512,7 @@ static __init void da850_evm_init(void) pr_warning("da850_evm_init: mcasp mux setup failed: %d\n", ret); + da850_evm_snd_data.sram_pool = sram_get_gen_pool(); da8xx_register_mcasp(0, &da850_evm_snd_data); ret = davinci_cfg_reg_list(da850_lcdcntl_pins); -- cgit v0.10.2 From d0c6c482f6d8c2145717ab0266c87e3e792527ef Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 5 Dec 2012 18:20:36 +0100 Subject: ASoC: McASP: remove unused variables codec_fmt and sample_rate variables are unused in both snd_platform_data and davinci_audio_dev, so drop them. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index f3d6e4f..8db5ae0 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -23,7 +23,6 @@ struct snd_platform_data { u32 rx_dma_offset; int asp_chan_q; /* event queue number for ASP channel */ int ram_chan_q; /* event queue number for RAM channel */ - unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps * caused by underruns, but will swap the left and right channels diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5715595..7c24837 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1158,7 +1158,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->tdm_slots = pdata->tdm_slots; dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; - dev->codec_fmt = pdata->codec_fmt; dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 156f15f..a42b5d9 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,7 @@ enum { struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; - int sample_rate; struct device *dev; - unsigned int codec_fmt; /* McASP specific data */ int tdm_slots; -- cgit v0.10.2 From ba764b3def0b979fd22884779f1a02f701e3af72 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 5 Dec 2012 18:20:37 +0100 Subject: ASoC: McASP: calculate values for channel size Change davinci_config_channel_size() to derive the values for XSSZ and XROT in DAVINCI_MCASP_[RT]XFMT_REG from the configured word length rather than hard-coding them in a switch/case block. Also, by directly passing the word length to davinci_config_channel_size(), we can get rid of the DAVINCI_AUDIO_WORD_* enum. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 7c24837..1b0aac9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -619,57 +619,14 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, } static int davinci_config_channel_size(struct davinci_audio_dev *dev, - int channel_size) + int word_length) { - u32 fmt = 0; - u32 mask, rotate; - - switch (channel_size) { - case DAVINCI_AUDIO_WORD_8: - fmt = 0x03; - rotate = 6; - mask = 0x000000ff; - break; - - case DAVINCI_AUDIO_WORD_12: - fmt = 0x05; - rotate = 5; - mask = 0x00000fff; - break; - - case DAVINCI_AUDIO_WORD_16: - fmt = 0x07; - rotate = 4; - mask = 0x0000ffff; - break; - - case DAVINCI_AUDIO_WORD_20: - fmt = 0x09; - rotate = 3; - mask = 0x000fffff; - break; - - case DAVINCI_AUDIO_WORD_24: - fmt = 0x0B; - rotate = 2; - mask = 0x00ffffff; - break; - - case DAVINCI_AUDIO_WORD_28: - fmt = 0x0D; - rotate = 1; - mask = 0x0fffffff; - break; - - case DAVINCI_AUDIO_WORD_32: - fmt = 0x0F; - rotate = 0; - mask = 0xffffffff; - break; + u32 fmt; + u32 rotate = (32 - word_length) / 4; + u32 mask = (1ULL << word_length) - 1; - default: - return -EINVAL; - } + /* mapping of the XSSZ bit-field as described in the datasheet */ + fmt = (word_length >> 1) - 1; mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), RXSSZ(0x0F)); @@ -856,19 +813,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: dma_params->data_type = 1; - word_length = DAVINCI_AUDIO_WORD_8; + word_length = 8; break; case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: dma_params->data_type = 2; - word_length = DAVINCI_AUDIO_WORD_16; + word_length = 16; break; case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: dma_params->data_type = 3; - word_length = DAVINCI_AUDIO_WORD_24; + word_length = 24; break; case SNDRV_PCM_FORMAT_U24_LE: @@ -876,7 +833,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; - word_length = DAVINCI_AUDIO_WORD_32; + word_length = 32; break; default: diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a42b5d9..d2449a8 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -27,16 +27,6 @@ #define DAVINCI_MCASP_I2S_DAI 0 #define DAVINCI_MCASP_DIT_DAI 1 -enum { - DAVINCI_AUDIO_WORD_8 = 0, - DAVINCI_AUDIO_WORD_12, - DAVINCI_AUDIO_WORD_16, - DAVINCI_AUDIO_WORD_20, - DAVINCI_AUDIO_WORD_24, - DAVINCI_AUDIO_WORD_32, - DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ -}; - struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; -- cgit v0.10.2 From 1b3bc060fb008ddd75fe60c876c24784a517c10c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 5 Dec 2012 18:20:38 +0100 Subject: ASoC: McASP: implement a way to force BCLK/LRCLK ratios Depending on the Codec, the the BCLK/LRCLK ratio might not be freely chosen by the CPU DAI. For example, some Codec might want to be supplied with 32-bit samples for both its channels regardless of the actual audio word size the CPU sends. In such cases, the rest of the bits on the data lines must be padded with zeros: _______________________________ LRCLK / \ --' `---------- ..... BCLK ||||||||||||||||||||||||||||||||||||||||||||||| ..... DATA ____||||||||||||||||_________________|||||||||| ..... |<-- data -->|<-- pads --> | This patch adds a new clock divider to configure the BCLK/LRCLK ratio. If the machine code uses that divider, the driver uses the specified value, instead of deriving that information from the audio word size. Otherwise, the original behaviour is retained. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1b0aac9..55e2bf6 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -593,6 +593,10 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div ACLKRDIV(div - 1), ACLKRDIV_MASK); break; + case 2: /* BCLK/LRCLK ratio */ + dev->bclk_lrclk_ratio = div; + break; + default: return -EINVAL; } @@ -625,6 +629,17 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, u32 rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() + * callback, take it into account here. That allows us to for example + * send 32 bits per channel to the codec, while only 16 of them carry + * audio payload. + * The clock ratio is given for a full period of data (both left and + * right channels), so it has to be divided by 2. + */ + if (dev->bclk_lrclk_ratio) + word_length = dev->bclk_lrclk_ratio / 2; + /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index d2449a8..0edd3b5 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -38,6 +38,7 @@ struct davinci_audio_dev { u8 num_serializer; u8 *serial_dir; u8 version; + u8 bclk_lrclk_ratio; /* McASP FIFO related */ u8 txnumevt; -- cgit v0.10.2