diff options
author | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2010-08-20 16:19:27 (GMT) |
---|---|---|
committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2010-08-20 16:19:27 (GMT) |
commit | bf557a50f59fc62dfd89fa5bf08c6f5d96fb2f45 (patch) | |
tree | 81a2cd45044970ccceec9ae4e1547e8f5dee850d /sound | |
parent | 26b01ccdc8ded270a1a65721b99acacf0c44e088 (diff) | |
parent | 3fabe089ad8b8f238bc9de3e7586ae8d2a81f57c (diff) | |
download | linux-fsl-qoriq-bf557a50f59fc62dfd89fa5bf08c6f5d96fb2f45.tar.xz |
Merge branch 'for-2.6.37' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.37
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/codecs/wl1273.c | 525 | ||||
-rw-r--r-- | sound/soc/codecs/wl1273.h | 101 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_dma.c | 67 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_ssi.c | 25 |
4 files changed, 699 insertions, 19 deletions
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c new file mode 100644 index 0000000..0cd5909 --- /dev/null +++ b/sound/soc/codecs/wl1273.c @@ -0,0 +1,525 @@ +/* + * ALSA SoC WL1273 codec driver + * + * Author: Matti Aaltonen, <matti.j.aaltonen@nokia.com> + * + * Copyright: (C) 2010 Nokia Corporation + * + * 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 <linux/mfd/wl1273-core.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wl1273.h" + +enum wl1273_mode { WL1273_MODE_BT, WL1273_MODE_FM_RX, WL1273_MODE_FM_TX }; + +/* codec private data */ +struct wl1273_priv { + enum wl1273_mode mode; + struct wl1273_core *core; + unsigned int channels; +}; + +static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, + int rate, int width) +{ + struct device *dev = &core->i2c_dev->dev; + int r = 0; + u16 mode; + + dev_dbg(dev, "rate: %d\n", rate); + dev_dbg(dev, "width: %d\n", width); + + mutex_lock(&core->lock); + + mode = core->i2s_mode & ~WL1273_IS2_WIDTH & ~WL1273_IS2_RATE; + + switch (rate) { + case 48000: + mode |= WL1273_IS2_RATE_48K; + break; + case 44100: + mode |= WL1273_IS2_RATE_44_1K; + break; + case 32000: + mode |= WL1273_IS2_RATE_32K; + break; + case 22050: + mode |= WL1273_IS2_RATE_22_05K; + break; + case 16000: + mode |= WL1273_IS2_RATE_16K; + break; + case 12000: + mode |= WL1273_IS2_RATE_12K; + break; + case 11025: + mode |= WL1273_IS2_RATE_11_025; + break; + case 8000: + mode |= WL1273_IS2_RATE_8K; + break; + default: + dev_err(dev, "Sampling rate: %d not supported\n", rate); + r = -EINVAL; + goto out; + } + + switch (width) { + case 16: + mode |= WL1273_IS2_WIDTH_32; + break; + case 20: + mode |= WL1273_IS2_WIDTH_40; + break; + case 24: + mode |= WL1273_IS2_WIDTH_48; + break; + case 25: + mode |= WL1273_IS2_WIDTH_50; + break; + case 30: + mode |= WL1273_IS2_WIDTH_60; + break; + case 32: + mode |= WL1273_IS2_WIDTH_64; + break; + case 40: + mode |= WL1273_IS2_WIDTH_80; + break; + case 48: + mode |= WL1273_IS2_WIDTH_96; + break; + case 64: + mode |= WL1273_IS2_WIDTH_128; + break; + default: + dev_err(dev, "Data width: %d not supported\n", width); + r = -EINVAL; + goto out; + } + + dev_dbg(dev, "WL1273_I2S_DEF_MODE: 0x%04x\n", WL1273_I2S_DEF_MODE); + dev_dbg(dev, "core->i2s_mode: 0x%04x\n", core->i2s_mode); + dev_dbg(dev, "mode: 0x%04x\n", mode); + + if (core->i2s_mode != mode) { + r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode); + if (r) + goto out; + + core->i2s_mode = mode; + r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE, + WL1273_AUDIO_ENABLE_I2S); + if (r) + goto out; + } +out: + mutex_unlock(&core->lock); + + return r; +} + +static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, + int channel_number) +{ + struct i2c_client *client = core->i2c_dev; + struct device *dev = &client->dev; + int r = 0; + + dev_dbg(dev, "%s\n", __func__); + + mutex_lock(&core->lock); + + if (core->channel_number == channel_number) + goto out; + + if (channel_number == 1 && core->mode == WL1273_MODE_RX) + r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, + WL1273_RX_MONO); + else if (channel_number == 1 && core->mode == WL1273_MODE_TX) + r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, + WL1273_TX_MONO); + else if (channel_number == 2 && core->mode == WL1273_MODE_RX) + r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, + WL1273_RX_STEREO); + else if (channel_number == 2 && core->mode == WL1273_MODE_TX) + r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, + WL1273_TX_STEREO); + else + r = -EINVAL; +out: + mutex_unlock(&core->lock); + + return r; +} + +static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = wl1273->mode; + + return 0; +} + +static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; + +static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + /* 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] >= ARRAY_SIZE(wl1273_audio_route)) + return -EINVAL; + + wl1273->mode = ucontrol->value.integer.value[0]; + + return 1; +} + +static const struct soc_enum wl1273_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route); + +static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + ucontrol->value.integer.value[0] = wl1273->core->audio_mode; + + return 0; +} + +static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + int val, r = 0; + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + val = ucontrol->value.integer.value[0]; + if (wl1273->core->audio_mode == val) + return 0; + + r = wl1273_fm_set_audio(wl1273->core, val); + if (r < 0) + return r; + + return 1; +} + +static const char *wl1273_audio_strings[] = { "Digital", "Analog" }; + +static const struct soc_enum wl1273_audio_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings), + wl1273_audio_strings); + +static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + ucontrol->value.integer.value[0] = wl1273->core->volume; + + return 0; +} + +static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + int r; + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + r = wl1273_fm_set_volume(wl1273->core, + ucontrol->value.integer.value[0]); + if (r) + return r; + + return 1; +} + +static const struct snd_kcontrol_new wl1273_controls[] = { + SOC_ENUM_EXT("Codec Mode", wl1273_enum, + snd_wl1273_get_audio_route, snd_wl1273_set_audio_route), + SOC_ENUM_EXT("Audio Switch", wl1273_audio_enum, + snd_wl1273_fm_audio_get, snd_wl1273_fm_audio_put), + SOC_SINGLE_EXT("Volume", 0, 0, WL1273_MAX_VOLUME, 0, + snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put), +}; + +static int wl1273_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + switch (wl1273->mode) { + case WL1273_MODE_BT: + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 8000, 8000); + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1); + break; + case WL1273_MODE_FM_RX: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + pr_err("Cannot play in RX mode.\n"); + return -EINVAL; + } + break; + case WL1273_MODE_FM_TX: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + pr_err("Cannot capture in TX mode.\n"); + return -EINVAL; + } + break; + default: + return -EINVAL; + break; + } + + return 0; +} + +static int wl1273_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 wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec); + struct wl1273_core *core = wl1273->core; + unsigned int rate, width, r; + + if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) { + pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n"); + return -EINVAL; + } + + rate = params_rate(params); + width = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + + if (wl1273->mode == WL1273_MODE_BT) { + if (rate != 8000) { + pr_err("Rate %d not supported.\n", params_rate(params)); + return -EINVAL; + } + + if (params_channels(params) != 1) { + pr_err("Only mono supported.\n"); + return -EINVAL; + } + + return 0; + } + + if (wl1273->mode == WL1273_MODE_FM_TX && + substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + pr_err("Only playback supported with TX.\n"); + return -EINVAL; + } + + if (wl1273->mode == WL1273_MODE_FM_RX && + substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + pr_err("Only capture supported with RX.\n"); + return -EINVAL; + } + + if (wl1273->mode != WL1273_MODE_FM_RX && + wl1273->mode != WL1273_MODE_FM_TX) { + pr_err("Unexpected mode: %d.\n", wl1273->mode); + return -EINVAL; + } + + r = snd_wl1273_fm_set_i2s_mode(core, rate, width); + if (r) + return r; + + wl1273->channels = params_channels(params); + r = snd_wl1273_fm_set_channel_number(core, wl1273->channels); + if (r) + return r; + + return 0; +} + +static struct snd_soc_dai_ops wl1273_dai_ops = { + .startup = wl1273_startup, + .hw_params = wl1273_hw_params, +}; + +static struct snd_soc_dai_driver wl1273_dai = { + .name = "wl1273-fm", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE}, + .ops = &wl1273_dai_ops, +}; + +/* Audio interface format for the soc_card driver */ +int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt) +{ + struct wl1273_priv *wl1273; + + if (codec == NULL || fmt == NULL) + return -EINVAL; + + wl1273 = snd_soc_codec_get_drvdata(codec); + + switch (wl1273->mode) { + case WL1273_MODE_FM_RX: + case WL1273_MODE_FM_TX: + *fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + break; + case WL1273_MODE_BT: + *fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + break; + default: + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(wl1273_get_format); + +static int wl1273_probe(struct snd_soc_codec *codec) +{ + struct wl1273_core **core = codec->dev->platform_data; + struct wl1273_priv *wl1273; + int r; + + dev_dbg(codec->dev, "%s.\n", __func__); + + if (!core) { + dev_err(codec->dev, "Platform data is missing.\n"); + return -EINVAL; + } + + wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL); + if (wl1273 == NULL) { + dev_err(codec->dev, "Cannot allocate memory.\n"); + return -ENOMEM; + } + + wl1273->mode = WL1273_MODE_BT; + wl1273->core = *core; + + snd_soc_codec_set_drvdata(codec, wl1273); + mutex_init(&codec->mutex); + + r = snd_soc_add_controls(codec, wl1273_controls, + ARRAY_SIZE(wl1273_controls)); + if (r) + kfree(wl1273); + + return r; +} + +static int wl1273_remove(struct snd_soc_codec *codec) +{ + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s\n", __func__); + kfree(wl1273); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { + .probe = wl1273_probe, + .remove = wl1273_remove, +}; + +static int __devinit wl1273_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wl1273, + &wl1273_dai, 1); +} + +static int __devexit wl1273_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +MODULE_ALIAS("platform:wl1273-codec"); + +static struct platform_driver wl1273_platform_driver = { + .driver = { + .name = "wl1273-codec", + .owner = THIS_MODULE, + }, + .probe = wl1273_platform_probe, + .remove = __devexit_p(wl1273_platform_remove), +}; + +static int __init wl1273_init(void) +{ + return platform_driver_register(&wl1273_platform_driver); +} +module_init(wl1273_init); + +static void __exit wl1273_exit(void) +{ + platform_driver_unregister(&wl1273_platform_driver); +} +module_exit(wl1273_exit); + +MODULE_AUTHOR("Matti Aaltonen <matti.j.aaltonen@nokia.com>"); +MODULE_DESCRIPTION("ASoC WL1273 codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h new file mode 100644 index 0000000..14ed027 --- /dev/null +++ b/sound/soc/codecs/wl1273.h @@ -0,0 +1,101 @@ +/* + * sound/soc/codec/wl1273.h + * + * ALSA SoC WL1273 codec driver + * + * Copyright (C) Nokia Corporation + * Author: Matti Aaltonen <matti.j.aaltonen@nokia.com> + * + * 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 __WL1273_CODEC_H__ +#define __WL1273_CODEC_H__ + +/* I2S protocol, left channel first, data width 16 bits */ +#define WL1273_PCM_DEF_MODE 0x00 + +/* Rx */ +#define WL1273_AUDIO_ENABLE_I2S (1 << 0) +#define WL1273_AUDIO_ENABLE_ANALOG (1 << 1) + +/* Tx */ +#define WL1273_AUDIO_IO_SET_ANALOG 0 +#define WL1273_AUDIO_IO_SET_I2S 1 + +#define WL1273_POWER_SET_OFF 0 +#define WL1273_POWER_SET_FM (1 << 0) +#define WL1273_POWER_SET_RDS (1 << 1) +#define WL1273_POWER_SET_RETENTION (1 << 4) + +#define WL1273_PUPD_SET_OFF 0x00 +#define WL1273_PUPD_SET_ON 0x01 +#define WL1273_PUPD_SET_RETENTION 0x10 + +/* I2S mode */ +#define WL1273_IS2_WIDTH_32 0x0 +#define WL1273_IS2_WIDTH_40 0x1 +#define WL1273_IS2_WIDTH_22_23 0x2 +#define WL1273_IS2_WIDTH_23_22 0x3 +#define WL1273_IS2_WIDTH_48 0x4 +#define WL1273_IS2_WIDTH_50 0x5 +#define WL1273_IS2_WIDTH_60 0x6 +#define WL1273_IS2_WIDTH_64 0x7 +#define WL1273_IS2_WIDTH_80 0x8 +#define WL1273_IS2_WIDTH_96 0x9 +#define WL1273_IS2_WIDTH_128 0xa +#define WL1273_IS2_WIDTH 0xf + +#define WL1273_IS2_FORMAT_STD (0x0 << 4) +#define WL1273_IS2_FORMAT_LEFT (0x1 << 4) +#define WL1273_IS2_FORMAT_RIGHT (0x2 << 4) +#define WL1273_IS2_FORMAT_USER (0x3 << 4) + +#define WL1273_IS2_MASTER (0x0 << 6) +#define WL1273_IS2_SLAVEW (0x1 << 6) + +#define WL1273_IS2_TRI_AFTER_SENDING (0x0 << 7) +#define WL1273_IS2_TRI_ALWAYS_ACTIVE (0x1 << 7) + +#define WL1273_IS2_SDOWS_RR (0x0 << 8) +#define WL1273_IS2_SDOWS_RF (0x1 << 8) +#define WL1273_IS2_SDOWS_FR (0x2 << 8) +#define WL1273_IS2_SDOWS_FF (0x3 << 8) + +#define WL1273_IS2_TRI_OPT (0x0 << 10) +#define WL1273_IS2_TRI_ALWAYS (0x1 << 10) + +#define WL1273_IS2_RATE_48K (0x0 << 12) +#define WL1273_IS2_RATE_44_1K (0x1 << 12) +#define WL1273_IS2_RATE_32K (0x2 << 12) +#define WL1273_IS2_RATE_22_05K (0x4 << 12) +#define WL1273_IS2_RATE_16K (0x5 << 12) +#define WL1273_IS2_RATE_12K (0x8 << 12) +#define WL1273_IS2_RATE_11_025 (0x9 << 12) +#define WL1273_IS2_RATE_8K (0xa << 12) +#define WL1273_IS2_RATE (0xf << 12) + +#define WL1273_I2S_DEF_MODE (WL1273_IS2_WIDTH_32 | \ + WL1273_IS2_FORMAT_STD | \ + WL1273_IS2_MASTER | \ + WL1273_IS2_TRI_AFTER_SENDING | \ + WL1273_IS2_SDOWS_RR | \ + WL1273_IS2_TRI_OPT | \ + WL1273_IS2_RATE_48K) + +int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt); + +#endif /* End of __WL1273_CODEC_H__ */ diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 5a6f56d..f039e8d 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -60,6 +60,7 @@ struct dma_object { struct snd_soc_platform_driver dai; dma_addr_t ssi_stx_phys; dma_addr_t ssi_srx_phys; + unsigned int ssi_fifo_depth; struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; @@ -99,6 +100,7 @@ struct fsl_dma_private { unsigned int irq; struct snd_pcm_substream *substream; dma_addr_t ssi_sxx_phys; + unsigned int ssi_fifo_depth; dma_addr_t ld_buf_phys; unsigned int current_link; dma_addr_t dma_buf_phys; @@ -439,6 +441,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) else dma_private->ssi_sxx_phys = dma->ssi_srx_phys; + dma_private->ssi_fifo_depth = dma->ssi_fifo_depth; dma_private->dma_channel = dma->channel; dma_private->irq = dma->irq; dma_private->substream = substream; @@ -552,11 +555,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, struct device *dev = rtd->platform->dev; /* Number of bits per sample */ - unsigned int sample_size = + unsigned int sample_bits = snd_pcm_format_physical_width(params_format(hw_params)); /* Number of bytes per frame */ - unsigned int frame_size = 2 * (sample_size / 8); + unsigned int sample_bytes = sample_bits / 8; /* Bus address of SSI STX register */ dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys; @@ -596,7 +599,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * that offset here. While we're at it, also tell the DMA controller * how much data to transfer per sample. */ - switch (sample_size) { + switch (sample_bits) { case 8: mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; ssi_sxx_phys += 3; @@ -610,22 +613,42 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, break; default: /* We should never get here */ - dev_err(dev, "unsupported sample size %u\n", sample_size); + dev_err(dev, "unsupported sample size %u\n", sample_bits); return -EINVAL; } /* - * BWC should always be a multiple of the frame size. BWC determines - * how many bytes are sent/received before the DMA controller checks the - * SSI to see if it needs to stop. For playback, the transmit FIFO can - * hold three frames, so we want to send two frames at a time. For - * capture, the receive FIFO is triggered when it contains one frame, so - * we want to receive one frame at a time. + * BWC determines how many bytes are sent/received before the DMA + * controller checks the SSI to see if it needs to stop. BWC should + * always be a multiple of the frame size, so that we always transmit + * whole frames. Each frame occupies two slots in the FIFO. The + * parameter for CCSR_DMA_MR_BWC() is rounded down the next power of two + * (MR[BWC] can only represent even powers of two). + * + * To simplify the process, we set BWC to the largest value that is + * less than or equal to the FIFO watermark. For playback, this ensures + * that we transfer the maximum amount without overrunning the FIFO. + * For capture, this ensures that we transfer the maximum amount without + * underrunning the FIFO. + * + * f = SSI FIFO depth + * w = SSI watermark value (which equals f - 2) + * b = DMA bandwidth count (in bytes) + * s = sample size (in bytes, which equals frame_size * 2) + * + * For playback, we never transmit more than the transmit FIFO + * watermark, otherwise we might write more data than the FIFO can hold. + * The watermark is equal to the FIFO depth minus two. + * + * For capture, two equations must hold: + * w > f - (b / s) + * w >= b / s + * + * So, b > 2 * s, but b must also be <= s * w. To simplify, we set + * b = s * w, which is equal to + * (dma_private->ssi_fifo_depth - 2) * sample_bytes. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - mr |= CCSR_DMA_MR_BWC(2 * frame_size); - else - mr |= CCSR_DMA_MR_BWC(frame_size); + mr |= CCSR_DMA_MR_BWC((dma_private->ssi_fifo_depth - 2) * sample_bytes); out_be32(&dma_channel->mr, mr); @@ -879,6 +902,7 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, struct device_node *np = of_dev->dev.of_node; struct device_node *ssi_np; struct resource res; + const uint32_t *iprop; int ret; /* Find the SSI node that points to us. */ @@ -889,15 +913,17 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, } ret = of_address_to_resource(ssi_np, 0, &res); - of_node_put(ssi_np); if (ret) { - dev_err(&of_dev->dev, "could not determine device resources\n"); + dev_err(&of_dev->dev, "could not determine resources for %s\n", + ssi_np->full_name); + of_node_put(ssi_np); return ret; } dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); if (!dma) { dev_err(&of_dev->dev, "could not allocate dma object\n"); + of_node_put(ssi_np); return -ENOMEM; } @@ -910,6 +936,15 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); + iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); + if (iprop) + dma->ssi_fifo_depth = *iprop; + else + /* Older 8610 DTs didn't have the fifo-depth property */ + dma->ssi_fifo_depth = 8; + + of_node_put(ssi_np); + ret = snd_soc_register_platform(&of_dev->dev, &dma->dai); if (ret) { dev_err(&of_dev->dev, "could not register platform\n"); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7939c33..d1c855a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -93,6 +93,7 @@ struct fsl_ssi_private { unsigned int playback; unsigned int capture; int asynchronous; + unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; struct platform_device *pdev; @@ -337,11 +338,20 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We - * don't use FIFO 1. Since the SSI only supports stereo, the - * watermark should never be an odd number. + * don't use FIFO 1. We program the transmit water to signal a + * DMA transfer if there are only two (or fewer) elements left + * in the FIFO. Two elements equals one frame (left channel, + * right channel). This value, however, depends on the depth of + * the transmit buffer. + * + * We program the receive FIFO to notify us if at least two + * elements (one frame) have been written to the FIFO. We could + * make this value larger (and maybe we should), but this way + * data will be written to memory as soon as it's available. */ out_be32(&ssi->sfcsr, - CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2)); + CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | + CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2)); /* * We keep the SSI disabled because if we enable it, then the @@ -622,6 +632,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, struct device_attribute *dev_attr = NULL; struct device_node *np = of_dev->dev.of_node; const char *p, *sprop; + const uint32_t *iprop; struct resource res; char name[64]; @@ -678,6 +689,14 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, else ssi_private->cpu_dai_drv.symmetric_rates = 1; + /* Determine the FIFO depth. */ + iprop = of_get_property(np, "fsl,fifo-depth", NULL); + if (iprop) + ssi_private->fifo_depth = *iprop; + else + /* Older 8610 DTs didn't have the fifo-depth property */ + ssi_private->fifo_depth = 8; + /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; dev_attr->attr.name = "statistics"; |