seeed-voicecard/seeed-voicecard.c

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2017-11-10 09:52:24 +01:00
/*
* SEEED voice card
*
* (C) Copyright 2017-2018
* Seeed Technology Co., Ltd. <www.seeedstudio.com>
*
* base on ASoC simple sound card support
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*
* Copyright (C) 2012 Renesas Solutions Corp.
* Kuninori Morimoto <kuninori.morimoto.gx@renesas.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.
*/
/* #undef DEBUG */
#include <linux/version.h>
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#include <linux/clk.h>
#include <linux/device.h>
#include <linux/gpio.h>
#include <linux/module.h>
#include <linux/of.h>
#include <linux/of_gpio.h>
#include <linux/platform_device.h>
#include <linux/string.h>
#include <sound/soc.h>
#include <sound/soc-dai.h>
#include <sound/simple_card_utils.h>
#include "ac10x.h"
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#define LINUX_VERSION_IS_GEQ(x1,x2,x3) (LINUX_VERSION_CODE >= KERNEL_VERSION(x1,x2,x3))
v5.13: Conditionally re-adding asoc_simple_parse_xxx() commit e25704f84ca2b586e8e65d1b2ab686205b3076fe Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Date: Mon Apr 12 08:52:13 2021 +0900 ASoC: simple-card-utils: remove asoc_simple_parse_xxx() ASoC is now supporting multi DAI, but, current simple-card / audio-graph are assuming fixed single DAI. Now, asoc_simple_parse_xxx() macro is assuming single DAI. To support multi-CPU/Codec, this patch unpack asoc_simple_parse_xxx() macro, and uses "&dai_link->cpus[i]" instead of "dai_link->cpus". $ git log --format=oneline v5.12..v5.13 -- include/sound/simple_card_utils.h f6fcc820e0c96664e2f21c0d6bb60630243ef36a ASoC: audio-graph: move audio_graph_remove() to simple-card-utils.c 1a456b1c6be13514a8fc5c1a99e6763f491d17e9 ASoC: audio-graph: move audio_graph_card_probe() to simple-card-utils.c 343e55e71877415a23372388b3e0c59a9bba42f6 ASoC: simple-card-utils: Increase maximum number of links to 128 fcfd763bef4ff7f6371790979a6ceac9c4ac425a ASoC: simple-card-utils: tidyup asoc_simple_parse_convert() 3919249e80995ed5f125f94d05fcb6171f79e732 ASoC: simple-card-utils: tidyup dev_dbg() to use 1 line 33cd6b191f1cdb5f332717a80ce26f661f53e924 ASoC: simple-card-utils: tidyup debug info for clock c826ec0391c83f06354a4ebb25c7b2480c18f33a ASoC: simple-card-utils: multi support at asoc_simple_canonicalize_cpu/platform() 9830d3e99f51fc1c1c6ab8be7778fd205af198ad ASoC: simple-card-utils: add simple_props_to_xxx() macro 40d8cbe70e71be170e0a4fe6ab112d9aaa9cfb18 ASoC: simple-card-utils: indicate missing CPU/Codec numbers for debug ac813c625ad5c3ee98a99e1b37659a0d85178978 ASoC: simple-card-utils: indicate dai_fmt if exist e25704f84ca2b586e8e65d1b2ab686205b3076fe ASoC: simple-card-utils: remove asoc_simple_parse_xxx() fafc05aadd4b6ce5c161135de9d3a653fc054543 ASoC: simple-card-utils: use for_each_prop_xxx() f899006d558546a8ee39c93f816eb3847c5bc6c0 ASoC: simple-card-utils: remove li->dais/li->conf 205eb17eddb473c3159743c7d3aaf68db37b7231 ASoC: simple-card-utils: share dummy DAI and reduce memory f2138aed231c88d5c4fa8d06aa15ad19685087c2 ASoC: simple-card-utils: enable flexible CPU/Codec/Platform 050c7950fd706fec229af9f30e8ce254cea9b675 ASoC: simple-card-utils: alloc dai_link information for CPU/Codec/Platform
2021-10-09 01:45:09 +02:00
#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,13,0)
#define asoc_simple_parse_clk_cpu(dev, node, dai_link, simple_dai) \
asoc_simple_parse_clk(dev, node, simple_dai, dai_link->cpus)
#define asoc_simple_parse_clk_codec(dev, node, dai_link, simple_dai) \
asoc_simple_parse_clk(dev, node, simple_dai, dai_link->codecs)
#define asoc_simple_parse_cpu(node, dai_link, is_single_link) \
asoc_simple_parse_dai(node, dai_link->cpus, is_single_link)
#define asoc_simple_parse_codec(node, dai_link) \
asoc_simple_parse_dai(node, dai_link->codecs, NULL)
#define asoc_simple_parse_platform(node, dai_link) \
asoc_simple_parse_dai(node, dai_link->platforms, NULL)
#endif
/*
* single codec:
* 0 - allow multi codec
* 1 - yes
*/
#define _SINGLE_CODEC 1
struct seeed_card_data {
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struct snd_soc_card snd_card;
struct seeed_dai_props {
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struct asoc_simple_dai cpu_dai;
struct asoc_simple_dai codec_dai;
struct snd_soc_dai_link_component cpus; /* single cpu */
struct snd_soc_dai_link_component codecs; /* single codec */
struct snd_soc_dai_link_component platforms;
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unsigned int mclk_fs;
} *dai_props;
unsigned int mclk_fs;
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unsigned channels_playback_default;
unsigned channels_playback_override;
unsigned channels_capture_default;
unsigned channels_capture_override;
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struct snd_soc_dai_link *dai_link;
#if CONFIG_AC10X_TRIG_LOCK
spinlock_t lock;
#endif
struct work_struct work_codec_clk;
#define TRY_STOP_MAX 3
int try_stop;
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};
struct seeed_card_info {
const char *name;
const char *card;
const char *codec;
const char *platform;
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unsigned int daifmt;
struct asoc_simple_dai cpu_dai;
struct asoc_simple_dai codec_dai;
};
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#define seeed_priv_to_card(priv) (&(priv)->snd_card)
#define seeed_priv_to_dev(priv) ((priv)->snd_card.dev)
#define seeed_priv_to_link(priv, i) ((priv)->snd_card.dai_link + (i))
#define seeed_priv_to_props(priv, i) ((priv)->dai_props + (i))
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#define DAI "sound-dai"
#define CELL "#sound-dai-cells"
#define PREFIX "seeed-voice-card,"
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static int seeed_voice_card_startup(struct snd_pcm_substream *substream)
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{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
struct seeed_dai_props *dai_props =
seeed_priv_to_props(priv, rtd->num);
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int ret;
ret = clk_prepare_enable(dai_props->cpu_dai.clk);
if (ret)
return ret;
ret = clk_prepare_enable(dai_props->codec_dai.clk);
if (ret)
clk_disable_unprepare(dai_props->cpu_dai.clk);
if (asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min) {
priv->channels_playback_default = asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min;
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}
if (asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min) {
priv->channels_capture_default = asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min;
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}
asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_override;
asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_override;
asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_override;
asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_override;
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return ret;
}
static void seeed_voice_card_shutdown(struct snd_pcm_substream *substream)
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{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
struct seeed_dai_props *dai_props =
seeed_priv_to_props(priv, rtd->num);
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asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_default;
asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_default;
asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_default;
asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_default;
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clk_disable_unprepare(dai_props->cpu_dai.clk);
clk_disable_unprepare(dai_props->codec_dai.clk);
}
static int seeed_voice_card_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
struct seeed_dai_props *dai_props =
seeed_priv_to_props(priv, rtd->num);
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unsigned int mclk, mclk_fs = 0;
int ret = 0;
if (priv->mclk_fs)
mclk_fs = priv->mclk_fs;
else if (dai_props->mclk_fs)
mclk_fs = dai_props->mclk_fs;
if (mclk_fs) {
mclk = params_rate(params) * mclk_fs;
ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
SND_SOC_CLOCK_IN);
if (ret && ret != -ENOTSUPP)
goto err;
ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk,
SND_SOC_CLOCK_OUT);
if (ret && ret != -ENOTSUPP)
goto err;
}
return 0;
err:
return ret;
}
#define _SET_CLOCK_CNT 2
Second/Alternative change to correct for 5.5+ change in trigger order The idea is the same as the previous attempt: calls ac101_trigger() just before set_clock(1). https://github.com/respeaker/seeed-voicecard/issues/290 6mics / linear 4 mics: fails to record against v5.10 kernel https://github.com/raspberrypi/linux/issues/4279 [regression] alsa system call blocks on record between 5.4.83 and 5.5.19 In v5.5, commit 4378f1fbe924054a09ff0d4e39e1a581b9245252 Author: Peter Ujfalusi <peter.ujfalusi@ti.com> Date: Fri Sep 27 10:16:46 2019 +0300 ASoC: soc-pcm: Use different sequence for start/stop trigger On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-04-27 02:35:13 +02:00
static int (* _set_clock[_SET_CLOCK_CNT])(int y_start_n_stop, struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai);
2018-02-05 03:45:55 +01:00
Second/Alternative change to correct for 5.5+ change in trigger order The idea is the same as the previous attempt: calls ac101_trigger() just before set_clock(1). https://github.com/respeaker/seeed-voicecard/issues/290 6mics / linear 4 mics: fails to record against v5.10 kernel https://github.com/raspberrypi/linux/issues/4279 [regression] alsa system call blocks on record between 5.4.83 and 5.5.19 In v5.5, commit 4378f1fbe924054a09ff0d4e39e1a581b9245252 Author: Peter Ujfalusi <peter.ujfalusi@ti.com> Date: Fri Sep 27 10:16:46 2019 +0300 ASoC: soc-pcm: Use different sequence for start/stop trigger On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-04-27 02:35:13 +02:00
int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int, struct snd_pcm_substream *, int, struct snd_soc_dai *)) {
if (! _set_clock[stream]) {
_set_clock[stream] = set_clock;
}
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return 0;
}
EXPORT_SYMBOL(seeed_voice_card_register_set_clock);
/*
* work_cb_codec_clk: clear audio codec inner clock.
*/
static void work_cb_codec_clk(struct work_struct *work)
{
struct seeed_card_data *priv = container_of(work, struct seeed_card_data, work_codec_clk);
int r = 0;
if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) {
Second/Alternative change to correct for 5.5+ change in trigger order The idea is the same as the previous attempt: calls ac101_trigger() just before set_clock(1). https://github.com/respeaker/seeed-voicecard/issues/290 6mics / linear 4 mics: fails to record against v5.10 kernel https://github.com/raspberrypi/linux/issues/4279 [regression] alsa system call blocks on record between 5.4.83 and 5.5.19 In v5.5, commit 4378f1fbe924054a09ff0d4e39e1a581b9245252 Author: Peter Ujfalusi <peter.ujfalusi@ti.com> Date: Fri Sep 27 10:16:46 2019 +0300 ASoC: soc-pcm: Use different sequence for start/stop trigger On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-04-27 02:35:13 +02:00
r = r || _set_clock[SNDRV_PCM_STREAM_CAPTURE](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
}
if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) {
Second/Alternative change to correct for 5.5+ change in trigger order The idea is the same as the previous attempt: calls ac101_trigger() just before set_clock(1). https://github.com/respeaker/seeed-voicecard/issues/290 6mics / linear 4 mics: fails to record against v5.10 kernel https://github.com/raspberrypi/linux/issues/4279 [regression] alsa system call blocks on record between 5.4.83 and 5.5.19 In v5.5, commit 4378f1fbe924054a09ff0d4e39e1a581b9245252 Author: Peter Ujfalusi <peter.ujfalusi@ti.com> Date: Fri Sep 27 10:16:46 2019 +0300 ASoC: soc-pcm: Use different sequence for start/stop trigger On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-04-27 02:35:13 +02:00
r = r || _set_clock[SNDRV_PCM_STREAM_PLAYBACK](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
}
if (r && priv->try_stop++ < TRY_STOP_MAX) {
if (0 != schedule_work(&priv->work_codec_clk)) {}
}
return;
}
static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
#if CONFIG_AC10X_TRIG_LOCK
unsigned long flags;
#endif
int ret = 0;
dev_dbg(rtd->card->dev, "%s() stream=%s cmd=%d play:%d, capt:%d\n",
__FUNCTION__, snd_pcm_stream_str(substream), cmd,
dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, dai->stream[SNDRV_PCM_STREAM_CAPTURE].active);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (cancel_work_sync(&priv->work_codec_clk) != 0) {}
#if CONFIG_AC10X_TRIG_LOCK
/* I know it will degrades performance, but I have no choice */
spin_lock_irqsave(&priv->lock, flags);
#endif
Second/Alternative change to correct for 5.5+ change in trigger order The idea is the same as the previous attempt: calls ac101_trigger() just before set_clock(1). https://github.com/respeaker/seeed-voicecard/issues/290 6mics / linear 4 mics: fails to record against v5.10 kernel https://github.com/raspberrypi/linux/issues/4279 [regression] alsa system call blocks on record between 5.4.83 and 5.5.19 In v5.5, commit 4378f1fbe924054a09ff0d4e39e1a581b9245252 Author: Peter Ujfalusi <peter.ujfalusi@ti.com> Date: Fri Sep 27 10:16:46 2019 +0300 ASoC: soc-pcm: Use different sequence for start/stop trigger On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-04-27 02:35:13 +02:00
if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) _set_clock[SNDRV_PCM_STREAM_CAPTURE](1, substream, cmd, dai);
if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) _set_clock[SNDRV_PCM_STREAM_PLAYBACK](1, substream, cmd, dai);
#if CONFIG_AC10X_TRIG_LOCK
spin_unlock_irqrestore(&priv->lock, flags);
#endif
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
/* capture channel resync, if overrun */
if (dai->stream[SNDRV_PCM_STREAM_CAPTURE].active && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
break;
}
/* interrupt environment */
if (in_irq() || in_nmi() || in_serving_softirq()) {
priv->try_stop = 0;
if (0 != schedule_work(&priv->work_codec_clk)) {
}
} else {
Second/Alternative change to correct for 5.5+ change in trigger order The idea is the same as the previous attempt: calls ac101_trigger() just before set_clock(1). https://github.com/respeaker/seeed-voicecard/issues/290 6mics / linear 4 mics: fails to record against v5.10 kernel https://github.com/raspberrypi/linux/issues/4279 [regression] alsa system call blocks on record between 5.4.83 and 5.5.19 In v5.5, commit 4378f1fbe924054a09ff0d4e39e1a581b9245252 Author: Peter Ujfalusi <peter.ujfalusi@ti.com> Date: Fri Sep 27 10:16:46 2019 +0300 ASoC: soc-pcm: Use different sequence for start/stop trigger On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-04-27 02:35:13 +02:00
if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) _set_clock[SNDRV_PCM_STREAM_CAPTURE](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) _set_clock[SNDRV_PCM_STREAM_PLAYBACK](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
}
break;
default:
ret = -EINVAL;
}
dev_dbg(rtd->card->dev, "%s() stream=%s cmd=%d play:%d, capt:%d;finished %d\n",
__FUNCTION__, snd_pcm_stream_str(substream), cmd,
dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, dai->stream[SNDRV_PCM_STREAM_CAPTURE].active, ret);
return ret;
}
static struct snd_soc_ops seeed_voice_card_ops = {
.startup = seeed_voice_card_startup,
.shutdown = seeed_voice_card_shutdown,
.hw_params = seeed_voice_card_hw_params,
.trigger = seeed_voice_card_trigger,
2017-11-10 09:52:24 +01:00
};
copy asoc_simple_card_parse_dai() from v5.4:sound/soc/generic/simple-card.c Towards the end of the series in asoc-v5.2: commit 8f7f298a333761a528e103cda3b42f3a416ad1ee Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Date: Wed Mar 20 13:56:36 2019 +0900 ASoC: simple-card-utils: separate asoc_simple_card_parse_dai() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph are using asoc_simple_card_parse_dai() which is different implementation. But, these are implemanted at simple-card-utils. It should be implemanted at each files. This patch separate these into each files. $ git log --format=oneline 0580dde59438686d60762b6da9229ebec693b94f^..ad11e59f52d6fc75037ac3cb66dc711b83c1bbf8 ad11e59f52d6fc75037ac3cb66dc711b83c1bbf8 ASoC: simple-card-utils: rename asoc_simple_card_xxx() to asoc_simple_() 8f7f298a333761a528e103cda3b42f3a416ad1ee ASoC: simple-card-utils: separate asoc_simple_card_parse_dai() 65a5056b21202eff7f54243e587183f4bb6ed352 ASoC: simple-card-utils: share asoc_simple_card_init_priv() 629f75440a68220a78aef9d8569831824890c47d ASoC: simple-card-utils: share asoc_simple_be_hw_params_fixup() ad934ca8010843482d61fda46786449a9bc99e10 ASoC: simple-card-utils: share asoc_simple_dai_init() f48dcbb6d47d870cf3a03f453c923dd262158c66 ASoC: simple-card-utils: share asoc_simple_hw_param() 686911b46fb5a08df142fe22b6c06dc6fbd3ba65 ASoC: simple-card-utils: share asoc_simple_shutdown() f38df5bf0c9cb905fa9d5abc86c3a00128cdbba5 ASoC: simple-card-utils: share asoc_simple_startup() e59289cda8dec0153fa396864c8ba8092ec3b80d ASoC: simple_card_utils: share common priv for simple-card/audio-graph 0580dde59438686d60762b6da9229ebec693b94f ASoC: simple-card-utils: add asoc_simple_debug_info()
2020-04-26 23:48:50 +02:00
static int asoc_simple_parse_dai(struct device_node *node,
struct snd_soc_dai_link_component *dlc,
int *is_single_link)
{
struct of_phandle_args args;
int ret;
if (!node)
return 0;
/*
* Get node via "sound-dai = <&phandle port>"
* it will be used as xxx_of_node on soc_bind_dai_link()
*/
ret = of_parse_phandle_with_args(node, DAI, CELL, 0, &args);
if (ret)
return ret;
/*
* FIXME
*
* Here, dlc->dai_name is pointer to CPU/Codec DAI name.
* If user unbinded CPU or Codec driver, but not for Sound Card,
* dlc->dai_name is keeping unbinded CPU or Codec
* driver's pointer.
*
* If user re-bind CPU or Codec driver again, ALSA SoC will try
* to rebind Card via snd_soc_try_rebind_card(), but because of
* above reason, it might can't bind Sound Card.
* Because Sound Card is pointing to released dai_name pointer.
*
* To avoid this rebind Card issue,
* 1) It needs to alloc memory to keep dai_name eventhough
* CPU or Codec driver was unbinded, or
* 2) user need to rebind Sound Card everytime
* if he unbinded CPU or Codec.
*/
ret = snd_soc_of_get_dai_name(node, &dlc->dai_name);
if (ret < 0)
return ret;
dlc->of_node = args.np;
if (is_single_link)
*is_single_link = !args.args_count;
return 0;
}
static int asoc_simple_init_dai(struct snd_soc_dai *dai,
struct asoc_simple_dai *simple_dai)
{
int ret;
if (!simple_dai)
return 0;
if (simple_dai->sysclk) {
ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk,
simple_dai->clk_direction);
if (ret && ret != -ENOTSUPP) {
dev_err(dai->dev, "simple-card: set_sysclk error\n");
return ret;
}
}
if (simple_dai->slots) {
ret = snd_soc_dai_set_bclk_ratio(dai,
simple_dai->slots *
simple_dai->slot_width);
if (ret && ret != -ENOTSUPP) {
dev_err(dai->dev, "simple-card: set_tdm_slot error\n");
return ret;
}
}
return 0;
}
#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,7,0)
v6.0: ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local commit 28086d05ada6d03daa886aad0e469854b811311c Author: Charles Keepax <ckeepax@opensource.cirrus.com> Date: Thu May 19 16:43:18 2022 +0100 ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local The helper function snd_soc_component_is_codec is based off the presence of the non_legacy_dai_naming flag. This isn't super robust as CPU side components may also specify this flag, and indeed the kernel already contains a couple that do. After componentisation there isn't really a totally robust solution to identifying what is a CODEC driver, without introducing a flag specifically for that purpose, and really the desirable direction to move in is that the distinction doesn't matter. This patch does two things to try to mitigate these problems. Firstly, now that all the other users of the helper function have been removed, it makes the helper function local to the driver rather, than being part of the core. This should help to discourage any new code from being created that depends on the CODEC driver distinction. Secondly, it updates the helper function itself to use the endianness flag rather than the non_legacy_dai_naming flag. The endianness flag is definitely invalid on a CPU side component, so it a more reliable indicator that the device is definitely a CODEC. The vast majority of buses require the CODEC to set the endianness flag, so the number of corner cases should be fairly minimal. It is worth noting that CODECs sending audio over SPI, or built into the CPU CODECs are potential corner cases, however the hope is that in most cases those types of devices do not consitute a simple audio card. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20220519154318.2153729-57-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
2022-10-30 23:45:19 +01:00
static inline int asoc_simple_component_is_codec(struct snd_soc_component *component)
{
return component->driver->endianness;
}
static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_component *component;
struct snd_soc_pcm_stream *params;
struct snd_pcm_hardware hw;
int i, ret, stream;
/* Only Codecs */
for_each_rtd_components(rtd, i, component) {
v6.0: ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local commit 28086d05ada6d03daa886aad0e469854b811311c Author: Charles Keepax <ckeepax@opensource.cirrus.com> Date: Thu May 19 16:43:18 2022 +0100 ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local The helper function snd_soc_component_is_codec is based off the presence of the non_legacy_dai_naming flag. This isn't super robust as CPU side components may also specify this flag, and indeed the kernel already contains a couple that do. After componentisation there isn't really a totally robust solution to identifying what is a CODEC driver, without introducing a flag specifically for that purpose, and really the desirable direction to move in is that the distinction doesn't matter. This patch does two things to try to mitigate these problems. Firstly, now that all the other users of the helper function have been removed, it makes the helper function local to the driver rather, than being part of the core. This should help to discourage any new code from being created that depends on the CODEC driver distinction. Secondly, it updates the helper function itself to use the endianness flag rather than the non_legacy_dai_naming flag. The endianness flag is definitely invalid on a CPU side component, so it a more reliable indicator that the device is definitely a CODEC. The vast majority of buses require the CODEC to set the endianness flag, so the number of corner cases should be fairly minimal. It is worth noting that CODECs sending audio over SPI, or built into the CPU CODECs are potential corner cases, however the hope is that in most cases those types of devices do not consitute a simple audio card. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20220519154318.2153729-57-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
2022-10-30 23:45:19 +01:00
if (!asoc_simple_component_is_codec(component))
return 0;
}
/* Assumes the capabilities are the same for all supported streams */
for (stream = 0; stream < 2; stream++) {
ret = snd_soc_runtime_calc_hw(rtd, &hw, stream);
if (ret == 0)
break;
}
if (ret < 0) {
dev_err(rtd->dev, "simple-card: no valid dai_link params\n");
return ret;
}
params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL);
if (!params)
return -ENOMEM;
params->formats = hw.formats;
params->rates = hw.rates;
params->rate_min = hw.rate_min;
params->rate_max = hw.rate_max;
params->channels_min = hw.channels_min;
params->channels_max = hw.channels_max;
#if LINUX_VERSION_CODE >= KERNEL_VERSION(6,4,0)
v6.4: switch to use c2c_params instead of params commit 99fddc1618ff64cc71bd51dbf83bd13e74778fe0 Merge: 72456c24c8357 1ea63f29c2771 Author: Mark Brown <broonie@kernel.org> Date: Wed Apr 5 16:35:09 2023 +0100 ASoC: clarify Codec2Codec params Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>: ASoC is supporting Codec2Codec, but its parameter name is "params" and "num_params" which are very unclear naming. This patch-set clarifies it by replacing to c2c_params / num_c2c_params. commit 1ea63f29c27712d6b9c45af67cd71299d849c5e3 Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Date: Sun Apr 2 23:00:17 2023 +0000 ASoC: soc.h: remove unused params/num_params No drivers are using params/num_params any more. Let's remove these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87iledc2ke.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org> commit e7a73b05542d82e209af450dd90b730255f6e775 Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Date: Sun Apr 2 23:00:07 2023 +0000 ASoC: samsung: switch to use c2c_params instead of params ASoC is now using c2c_params instead of params. This patch replace it. num_c2c_params (was num_params) was not mandatory before, but let's set it by this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87jzytc2kp.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org> commit 433f4a1697fae78c34377de1ef3abd26aec8214e Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Date: Sun Apr 2 22:59:57 2023 +0000 ASoC: meson: switch to use c2c_params instead of params ASoC is now using c2c_params instead of params. This patch replace it. num_c2c_params (was num_params) was not mandatory before, but let's set it by this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87lej9c2ky.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org> commit 7ddc7f91beb285246e926e3adf0b292b071aea33 Author: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Date: Sun Apr 2 22:59:35 2023 +0000 ASoC: soc.h: clarify Codec2Codec params snd_soc_dai_link has params/num_params, but it is unclear that params for what. This patch clarify it is params for Codec2Codec. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87o7o5c2lk.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2023-07-27 02:52:33 +02:00
dai_link->c2c_params = params;
dai_link->num_c2c_params = 1;
#else
/* apparently this goes back to 5.6.x */
dai_link->params = params;
dai_link->num_params = 1;
#endif
return 0;
}
#endif
static int seeed_voice_card_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *codec = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_dai *cpu = asoc_rtd_to_cpu(rtd, 0);
struct seeed_dai_props *dai_props =
seeed_priv_to_props(priv, rtd->num);
int ret;
ret = asoc_simple_init_dai(codec, &dai_props->codec_dai);
if (ret < 0)
return ret;
ret = asoc_simple_init_dai(cpu, &dai_props->cpu_dai);
if (ret < 0)
return ret;
#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,7,0)
ret = asoc_simple_init_dai_link_params(rtd);
if (ret < 0)
return ret;
#endif
2021-04-13 02:18:51 +02:00
dev_dbg(rtd->card->dev, "codec \"%s\" mapping to cpu \"%s\"\n", codec->name, cpu->name);
return 0;
}
static int seeed_voice_card_dai_link_of(struct device_node *node,
struct seeed_card_data *priv,
2017-11-10 09:52:24 +01:00
int idx,
bool is_top_level_node)
{
struct device *dev = seeed_priv_to_dev(priv);
struct snd_soc_dai_link *dai_link = seeed_priv_to_link(priv, idx);
struct seeed_dai_props *dai_props = seeed_priv_to_props(priv, idx);
2017-11-10 09:52:24 +01:00
struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai;
struct asoc_simple_dai *codec_dai = &dai_props->codec_dai;
struct device_node *cpu = NULL;
struct device_node *plat = NULL;
struct device_node *codec = NULL;
char prop[128];
char *prefix = "";
int ret, single_cpu;
/* For single DAI link & old style of DT node */
if (is_top_level_node)
prefix = PREFIX;
snprintf(prop, sizeof(prop), "%scpu", prefix);
cpu = of_get_child_by_name(node, prop);
if (!cpu) {
ret = -EINVAL;
dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
goto dai_link_of_err;
}
2017-11-10 09:52:24 +01:00
snprintf(prop, sizeof(prop), "%splat", prefix);
plat = of_get_child_by_name(node, prop);
snprintf(prop, sizeof(prop), "%scodec", prefix);
codec = of_get_child_by_name(node, prop);
if (!codec) {
2017-11-10 09:52:24 +01:00
ret = -EINVAL;
dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
goto dai_link_of_err;
}
ret = asoc_simple_parse_daifmt(dev, node, codec,
2017-11-10 09:52:24 +01:00
prefix, &dai_link->dai_fmt);
if (ret < 0)
goto dai_link_of_err;
of_property_read_u32(node, "mclk-fs", &dai_props->mclk_fs);
ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu);
2017-11-10 09:52:24 +01:00
if (ret < 0)
goto dai_link_of_err;
#if _SINGLE_CODEC
ret = asoc_simple_parse_codec(codec, dai_link);
2017-11-10 09:52:24 +01:00
if (ret < 0)
goto dai_link_of_err;
#else
ret = snd_soc_of_get_dai_link_codecs(dev, codec, dai_link);
if (ret < 0) {
dev_err(dev, "parse codec info error %d\n", ret);
goto dai_link_of_err;
}
dev_dbg(dev, "dai_link num_codecs = %d\n", dai_link->num_codecs);
#endif
2017-11-10 09:52:24 +01:00
ret = asoc_simple_parse_platform(plat, dai_link);
2017-11-10 09:52:24 +01:00
if (ret < 0)
goto dai_link_of_err;
ret = snd_soc_of_parse_tdm_slot(cpu, &cpu_dai->tx_slot_mask,
&cpu_dai->rx_slot_mask,
&cpu_dai->slots,
&cpu_dai->slot_width);
dev_dbg(dev, "cpu_dai : slot,width,tx,rx = %d,%d,%d,%d\n",
cpu_dai->slots, cpu_dai->slot_width,
cpu_dai->tx_slot_mask, cpu_dai->rx_slot_mask
);
2017-11-10 09:52:24 +01:00
if (ret < 0)
goto dai_link_of_err;
ret = snd_soc_of_parse_tdm_slot(codec, &codec_dai->tx_slot_mask,
&codec_dai->rx_slot_mask,
&codec_dai->slots,
&codec_dai->slot_width);
if (ret < 0)
goto dai_link_of_err;
#if LINUX_VERSION_CODE <= KERNEL_VERSION(4,10,0)
ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai);
#else
ret = asoc_simple_parse_clk_cpu(dev, cpu, dai_link, cpu_dai);
#endif
2017-11-10 09:52:24 +01:00
if (ret < 0)
goto dai_link_of_err;
#if LINUX_VERSION_CODE <= KERNEL_VERSION(4,10,0)
ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai);
#else
ret = asoc_simple_parse_clk_codec(dev, codec, dai_link, codec_dai);
#endif
2017-11-10 09:52:24 +01:00
if (ret < 0)
goto dai_link_of_err;
ret = asoc_simple_set_dailink_name(dev, dai_link,
2017-11-10 09:52:24 +01:00
"%s-%s",
dai_link->cpus->dai_name,
#if _SINGLE_CODEC
dai_link->codecs->dai_name
#else
dai_link->codecs[0].dai_name
#endif
);
2017-11-10 09:52:24 +01:00
if (ret < 0)
goto dai_link_of_err;
dai_link->ops = &seeed_voice_card_ops;
dai_link->init = seeed_voice_card_dai_init;
2017-11-10 09:52:24 +01:00
dev_dbg(dev, "\tname : %s\n", dai_link->stream_name);
dev_dbg(dev, "\tformat : %04x\n", dai_link->dai_fmt);
dev_dbg(dev, "\tcpu : %s / %d\n",
dai_link->cpus->dai_name,
2017-11-10 09:52:24 +01:00
dai_props->cpu_dai.sysclk);
dev_dbg(dev, "\tcodec : %s / %d\n",
#if _SINGLE_CODEC
dai_link->codecs->dai_name,
#else
dai_link->codecs[0].dai_name,
#endif
2017-11-10 09:52:24 +01:00
dai_props->codec_dai.sysclk);
#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,13,0)
asoc_simple_canonicalize_cpu(dai_link->cpus, single_cpu);
#if _SINGLE_CODEC
asoc_simple_canonicalize_platform(dai_link->platforms, dai_link->cpus);
#endif
#else
asoc_simple_canonicalize_cpu(dai_link, single_cpu);
#if _SINGLE_CODEC
asoc_simple_canonicalize_platform(dai_link);
#endif
#endif
2017-11-10 09:52:24 +01:00
dai_link_of_err:
of_node_put(cpu);
of_node_put(codec);
return ret;
}
static int seeed_voice_card_parse_aux_devs(struct device_node *node,
struct seeed_card_data *priv)
2017-11-10 09:52:24 +01:00
{
struct device *dev = seeed_priv_to_dev(priv);
2017-11-10 09:52:24 +01:00
struct device_node *aux_node;
int i, n, len;
if (!of_find_property(node, PREFIX "aux-devs", &len))
return 0; /* Ok to have no aux-devs */
n = len / sizeof(__be32);
if (n <= 0)
return -EINVAL;
priv->snd_card.aux_dev = devm_kzalloc(dev,
n * sizeof(*priv->snd_card.aux_dev), GFP_KERNEL);
if (!priv->snd_card.aux_dev)
return -ENOMEM;
for (i = 0; i < n; i++) {
aux_node = of_parse_phandle(node, PREFIX "aux-devs", i);
if (!aux_node)
return -EINVAL;
priv->snd_card.aux_dev[i].dlc.of_node = aux_node;
2017-11-10 09:52:24 +01:00
}
priv->snd_card.num_aux_devs = n;
return 0;
}
static int seeed_voice_card_parse_of(struct device_node *node,
struct seeed_card_data *priv)
2017-11-10 09:52:24 +01:00
{
struct device *dev = seeed_priv_to_dev(priv);
2017-11-10 09:52:24 +01:00
struct device_node *dai_link;
int ret;
if (!node)
return -EINVAL;
dai_link = of_get_child_by_name(node, PREFIX "dai-link");
/* The off-codec widgets */
if (of_property_read_bool(node, PREFIX "widgets")) {
ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card,
PREFIX "widgets");
if (ret)
goto card_parse_end;
}
/* DAPM routes */
if (of_property_read_bool(node, PREFIX "routing")) {
ret = snd_soc_of_parse_audio_routing(&priv->snd_card,
PREFIX "routing");
if (ret)
goto card_parse_end;
}
/* Factor to mclk, used in hw_params() */
of_property_read_u32(node, PREFIX "mclk-fs", &priv->mclk_fs);
/* Single/Muti DAI link(s) & New style of DT node */
if (dai_link) {
struct device_node *np = NULL;
int i = 0;
for_each_child_of_node(node, np) {
dev_dbg(dev, "\tlink %d:\n", i);
ret = seeed_voice_card_dai_link_of(np, priv,
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i, false);
if (ret < 0) {
of_node_put(np);
goto card_parse_end;
}
i++;
}
} else {
/* For single DAI link & old style of DT node */
ret = seeed_voice_card_dai_link_of(node, priv, 0, true);
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if (ret < 0)
goto card_parse_end;
}
ret = asoc_simple_parse_card_name(&priv->snd_card, PREFIX);
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if (ret < 0)
goto card_parse_end;
ret = seeed_voice_card_parse_aux_devs(node, priv);
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priv->channels_playback_default = 0;
priv->channels_playback_override = 2;
priv->channels_capture_default = 0;
priv->channels_capture_override = 2;
of_property_read_u32(node, PREFIX "channels-playback-default",
&priv->channels_playback_default);
of_property_read_u32(node, PREFIX "channels-playback-override",
&priv->channels_playback_override);
of_property_read_u32(node, PREFIX "channels-capture-default",
&priv->channels_capture_default);
of_property_read_u32(node, PREFIX "channels-capture-override",
&priv->channels_capture_override);
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card_parse_end:
of_node_put(dai_link);
return ret;
}
#ifdef DEBUG
inline void seeed_debug_dai(struct seeed_card_data *priv,
char *name,
struct asoc_simple_dai *dai)
{
struct device *dev = seeed_priv_to_dev(priv);
if (dai->name)
dev_dbg(dev, "%s dai name = %s\n",
name, dai->name);
if (dai->sysclk)
dev_dbg(dev, "%s sysclk = %d\n",
name, dai->sysclk);
dev_dbg(dev, "%s direction = %s\n",
name, dai->clk_direction ? "OUT" : "IN");
if (dai->slots)
dev_dbg(dev, "%s slots = %d\n", name, dai->slots);
if (dai->slot_width)
dev_dbg(dev, "%s slot width = %d\n", name, dai->slot_width);
if (dai->tx_slot_mask)
dev_dbg(dev, "%s tx slot mask = %d\n", name, dai->tx_slot_mask);
if (dai->rx_slot_mask)
dev_dbg(dev, "%s rx slot mask = %d\n", name, dai->rx_slot_mask);
if (dai->clk)
dev_dbg(dev, "%s clk %luHz\n", name, clk_get_rate(dai->clk));
}
inline void seeed_debug_info(struct seeed_card_data *priv)
{
struct snd_soc_card *card = seeed_priv_to_card(priv);
struct device *dev = seeed_priv_to_dev(priv);
int i;
if (card->name)
dev_dbg(dev, "Card Name: %s\n", card->name);
for (i = 0; i < card->num_links; i++) {
struct seeed_dai_props *props = seeed_priv_to_props(priv, i);
struct snd_soc_dai_link *link = seeed_priv_to_link(priv, i);
dev_dbg(dev, "DAI%d\n", i);
seeed_debug_dai(priv, "cpu", &props->cpu_dai);
seeed_debug_dai(priv, "codec", &props->codec_dai);
if (link->name)
dev_dbg(dev, "dai name = %s\n", link->name);
dev_dbg(dev, "dai format = %04x\n", link->dai_fmt);
/*
if (props->adata.convert_rate)
dev_dbg(dev, "convert_rate = %d\n",
props->adata.convert_rate);
if (props->adata.convert_channels)
dev_dbg(dev, "convert_channels = %d\n",
props->adata.convert_channels);
if (props->codec_conf && props->codec_conf->name_prefix)
dev_dbg(dev, "name prefix = %s\n",
props->codec_conf->name_prefix);
*/
if (props->mclk_fs)
dev_dbg(dev, "mclk-fs = %d\n",
props->mclk_fs);
}
}
#else
#define seeed_debug_info(priv)
#endif /* DEBUG */
static int seeed_voice_card_probe(struct platform_device *pdev)
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{
struct seeed_card_data *priv;
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struct snd_soc_dai_link *dai_link;
struct seeed_dai_props *dai_props;
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struct device_node *np = pdev->dev.of_node;
struct device *dev = &pdev->dev;
int num, ret, i;
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/* Get the number of DAI links */
if (np && of_get_child_by_name(np, PREFIX "dai-link"))
num = of_get_child_count(np);
else
num = 1;
/* Allocate the private data and the DAI link array */
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
dai_props = devm_kzalloc(dev, sizeof(*dai_props) * num, GFP_KERNEL);
dai_link = devm_kzalloc(dev, sizeof(*dai_link) * num, GFP_KERNEL);
if (!dai_props || !dai_link)
return -ENOMEM;
/*
* Use snd_soc_dai_link_component instead of legacy style
* It is codec only. but cpu/platform will be supported in the future.
* see
* soc-core.c :: snd_soc_init_multicodec()
*
* "platform" might be removed
* see
* simple-card-utils.c :: asoc_simple_canonicalize_platform()
*/
for (i = 0; i < num; i++) {
dai_link[i].cpus = &dai_props[i].cpus;
dai_link[i].num_cpus = 1;
dai_link[i].codecs = &dai_props[i].codecs;
dai_link[i].num_codecs = 1;
dai_link[i].platforms = &dai_props[i].platforms;
dai_link[i].num_platforms = 1;
}
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priv->dai_props = dai_props;
priv->dai_link = dai_link;
/* Init snd_soc_card */
priv->snd_card.owner = THIS_MODULE;
priv->snd_card.dev = dev;
priv->snd_card.dai_link = priv->dai_link;
priv->snd_card.num_links = num;
if (np && of_device_is_available(np)) {
ret = seeed_voice_card_parse_of(np, priv);
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if (ret < 0) {
if (ret != -EPROBE_DEFER)
dev_err(dev, "parse error %d\n", ret);
goto err;
}
} else {
struct seeed_card_info *cinfo;
struct snd_soc_dai_link_component *cpus;
struct snd_soc_dai_link_component *codecs;
struct snd_soc_dai_link_component *platform;
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cinfo = dev->platform_data;
if (!cinfo) {
dev_err(dev, "no info for seeed-voice-card\n");
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return -EINVAL;
}
if (!cinfo->name ||
!cinfo->codec_dai.name ||
!cinfo->codec ||
!cinfo->platform ||
!cinfo->cpu_dai.name) {
dev_err(dev, "insufficient seeed_voice_card_info settings\n");
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return -EINVAL;
}
cpus = dai_link->cpus;
cpus->dai_name = cinfo->cpu_dai.name;
codecs = dai_link->codecs;
codecs->name = cinfo->codec;
codecs->dai_name = cinfo->codec_dai.name;
platform = dai_link->platforms;
platform->name = cinfo->platform;
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priv->snd_card.name = (cinfo->card) ? cinfo->card : cinfo->name;
dai_link->name = cinfo->name;
dai_link->stream_name = cinfo->name;
dai_link->dai_fmt = cinfo->daifmt;
dai_link->init = seeed_voice_card_dai_init;
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memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
sizeof(priv->dai_props->cpu_dai));
memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai,
sizeof(priv->dai_props->codec_dai));
}
snd_soc_card_set_drvdata(&priv->snd_card, priv);
#if CONFIG_AC10X_TRIG_LOCK
spin_lock_init(&priv->lock);
#endif
INIT_WORK(&priv->work_codec_clk, work_cb_codec_clk);
seeed_debug_info(priv);
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ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
if (ret >= 0)
return ret;
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err:
asoc_simple_clean_reference(&priv->snd_card);
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return ret;
}
static int seeed_voice_card_remove(struct platform_device *pdev)
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{
struct snd_soc_card *card = platform_get_drvdata(pdev);
struct seeed_card_data *priv = snd_soc_card_get_drvdata(card);
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if (cancel_work_sync(&priv->work_codec_clk) != 0) {
}
asoc_simple_clean_reference(card);
return 0;
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}
static const struct of_device_id seeed_voice_of_match[] = {
{ .compatible = "seeed-voicecard", },
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{},
};
MODULE_DEVICE_TABLE(of, seeed_voice_of_match);
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static struct platform_driver seeed_voice_card = {
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.driver = {
.name = "seeed-voicecard",
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.pm = &snd_soc_pm_ops,
.of_match_table = seeed_voice_of_match,
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},
.probe = seeed_voice_card_probe,
.remove = seeed_voice_card_remove,
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};
module_platform_driver(seeed_voice_card);
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MODULE_ALIAS("platform:seeed-voice-card");
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MODULE_LICENSE("GPL v2");
MODULE_DESCRIPTION("ASoC SEEED Voice Card");
MODULE_AUTHOR("PeterYang<linsheng.yang@seeed.cc>");