June 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCHv2] ASoC: core - Separate out PCM operations into new file.
by Liam Girdwood 09 Jun '11

09 Jun '11

In preparation for Dynamic PCM support (AKA DSP support). There will be future patches that add support to allow PCMs to be dynamically routed to multiple DAIs at startup and also during stream runtime. This patch moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will in simplify the ASoC core features into distinct files. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 612 +----------------------------------------------- sound/soc/soc-pcm.c | 639 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 641 insertions(+), 612 deletions(-) create mode 100644 sound/soc/soc-pcm.c diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1ed61c5..adb5719 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o soc-pcm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index df5608d..32d7d2f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -44,7 +44,6 @@ #define NAME_SIZE 32 -static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS @@ -58,7 +57,7 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -485,552 +484,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && - !rtd->dai_link->symmetric_rates) - return 0; - - /* This can happen if multiple streams are starting simultaneously - - * the second can need to get its constraints before the first has - * picked a rate. Complain and allow the application to carry on. - */ - if (!rtd->rate) { - dev_warn(&rtd->dev, - "Not enforcing symmetric_rates due to race\n"); - return 0; - } - - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } - - return 0; -} - -/* - * Called by ALSA when a PCM substream is opened, the runtime->hw record is - * then initialized and any private data can be allocated. This also calls - * startup for the cpu DAI, platform, machine and codec DAI. - */ -static int soc_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; - - mutex_lock(&pcm_mutex); - - /* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { - ret = cpu_dai->driver->ops->startup(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->open) { - ret = platform->driver->ops->open(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); - goto platform_err; - } - } - - if (codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); - goto codec_dai_err; - } - } - - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { - ret = rtd->dai_link->ops->startup(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); - goto machine_err; - } - } - - /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = - max(codec_dai_drv->playback.rate_min, - cpu_dai_drv->playback.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->playback.rate_max, - cpu_dai_drv->playback.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->playback.channels_min, - cpu_dai_drv->playback.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->playback.channels_max, - cpu_dai_drv->playback.channels_max); - runtime->hw.formats = - codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; - runtime->hw.rates = - codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; - if (codec_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->playback.rates; - if (cpu_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->playback.rates; - } else { - runtime->hw.rate_min = - max(codec_dai_drv->capture.rate_min, - cpu_dai_drv->capture.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->capture.rate_max, - cpu_dai_drv->capture.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->capture.channels_min, - cpu_dai_drv->capture.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->capture.channels_max, - cpu_dai_drv->capture.channels_max); - runtime->hw.formats = - codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; - runtime->hw.rates = - codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; - if (codec_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->capture.rates; - if (cpu_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->capture.rates; - } - - ret = -EINVAL; - snd_pcm_limit_hw_rates(runtime); - if (!runtime->hw.rates) { - printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.formats) { - printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.channels_min || !runtime->hw.channels_max || - runtime->hw.channels_min > runtime->hw.channels_max) { - printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - - /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); - if (ret != 0) - goto config_err; - } - - pr_debug("asoc: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); - pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); - pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, - runtime->hw.channels_max); - pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, - runtime->hw.rate_max); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active++; - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - codec_dai->capture_active++; - } - cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; - mutex_unlock(&pcm_mutex); - return 0; - -config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - -machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - -codec_dai_err: - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - -platform_err: - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Power down the audio subsystem pmdown_time msecs after close is called. - * This is to ensure there are no pops or clicks in between any music tracks - * due to DAPM power cycling. - */ -static void close_delayed_work(struct work_struct *work) -{ - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - mutex_lock(&pcm_mutex); - - pr_debug("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->driver->playback.stream_name, - codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); - - /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); -} - -/* - * Called by ALSA when a PCM substream is closed. Private data can be - * freed here. The cpu DAI, codec DAI, machine and platform are also - * shutdown. - */ -static int soc_codec_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active--; - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - codec_dai->capture_active--; - } - - cpu_dai->active--; - codec_dai->active--; - codec->active--; - - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); - - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); - - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - cpu_dai->runtime = NULL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); - } else { - /* capture streams can be powered down now */ - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); - return 0; -} - -/* - * Called by ALSA when the PCM substream is prepared, can set format, sample - * rate, etc. This function is non atomic and can be called multiple times, - * it can refer to the runtime info. - */ -static int soc_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { - ret = rtd->dai_link->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: machine prepare error\n"); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->prepare) { - ret = platform->driver->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: platform prepare error\n"); - goto out; - } - } - - if (codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: codec DAI prepare error\n"); - goto out; - } - } - - if (cpu_dai->driver->ops->prepare) { - ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: cpu DAI prepare error\n"); - goto out; - } - } - - /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; - cancel_delayed_work(&rtd->delayed_work); - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_START); - - snd_soc_dai_digital_mute(codec_dai, 0); - -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Called by ALSA when the hardware params are set by application. This - * function can also be called multiple times and can allocate buffers - * (using snd_pcm_lib_* ). It's non-atomic. - */ -static int soc_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { - ret = rtd->dai_link->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: machine hw_params failed\n"); - goto out; - } - } - - if (codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); - goto codec_err; - } - } - - if (cpu_dai->driver->ops->hw_params) { - ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); - goto interface_err; - } - } - - if (platform->driver->ops && platform->driver->ops->hw_params) { - ret = platform->driver->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); - goto platform_err; - } - } - - rtd->rate = params_rate(params); - -out: - mutex_unlock(&pcm_mutex); - return ret; - -platform_err: - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - -interface_err: - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - -codec_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Frees resources allocated by hw_params, can be called multiple times - */ -static int soc_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - /* apply codec digital mute */ - if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); - - /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - /* free any DMA resources */ - if (platform->driver->ops && platform->driver->ops->hw_free) - platform->driver->ops->hw_free(substream); - - /* now free hw params for the DAIs */ - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - - mutex_unlock(&pcm_mutex); - return 0; -} - -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; - } - - if (platform->driver->ops && platform->driver->ops->trigger) { - ret = platform->driver->ops->trigger(substream, cmd); - if (ret < 0) - return ret; - } - - if (cpu_dai->driver->ops->trigger) { - ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; - } - return 0; -} - -/* - * soc level wrapper for pointer callback - * If cpu_dai, codec_dai, platform driver has the delay callback, than - * the runtime->delay will be updated accordingly. - */ -static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t offset = 0; - snd_pcm_sframes_t delay = 0; - - if (platform->driver->ops && platform->driver->ops->pointer) - offset = platform->driver->ops->pointer(substream); - - if (cpu_dai->driver->ops->delay) - delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - - if (codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); - - if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); - - runtime->delay = delay; - - return offset; -} - -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_codec_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - #ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ int snd_soc_suspend(struct device *dev) @@ -1692,9 +1145,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (order != SND_SOC_COMP_ORDER_LAST) return 0; - /* DAPM dai link stream work */ - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); - ret = soc_post_component_init(card, codec, num, 0); if (ret) return ret; @@ -2114,66 +1564,6 @@ static struct platform_driver soc_driver = { .remove = soc_remove, }; -/* create a new pcm */ -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_pcm *pcm; - char new_name[64]; - int ret = 0, playback = 0, capture = 0; - - /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - - if (codec_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min) - capture = 1; - - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); - ret = snd_pcm_new(rtd->card->snd_card, new_name, - num, playback, capture, &pcm); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); - return ret; - } - - rtd->pcm = pcm; - pcm->private_data = rtd; - if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; - } - - if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); - - if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - - if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd); - if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); - return ret; - } - } - - pcm->private_free = platform->driver->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); - return ret; -} - /** * snd_soc_codec_volatile_register: Report if a register is volatile. * diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c new file mode 100644 index 0000000..9bebee8 --- /dev/null +++ b/sound/soc/soc-pcm.c @@ -0,0 +1,639 @@ +/* + * soc-pcm.c -- ALSA SoC PCM + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Authors: Liam Girdwood <lrg(a)ti.com> + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> + +static DEFINE_MUTEX(pcm_mutex); + +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (!codec_dai->driver->symmetric_rates && + !cpu_dai->driver->symmetric_rates && + !rtd->dai_link->symmetric_rates) + return 0; + + /* This can happen if multiple streams are starting simultaneously - + * the second can need to get its constraints before the first has + * picked a rate. Complain and allow the application to carry on. + */ + if (!rtd->rate) { + dev_warn(&rtd->dev, + "Not enforcing symmetric_rates due to race\n"); + return 0; + } + + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + rtd->rate, rtd->rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + + return 0; +} + +/* + * Called by ALSA when a PCM substream is opened, the runtime->hw record is + * then initialized and any private data can be allocated. This also calls + * startup for the cpu DAI, platform, machine and codec DAI. + */ +static int soc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; + int ret = 0; + + mutex_lock(&pcm_mutex); + + /* startup the audio subsystem */ + if (cpu_dai->driver->ops->startup) { + ret = cpu_dai->driver->ops->startup(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open interface %s\n", + cpu_dai->name); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->open) { + ret = platform->driver->ops->open(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + goto platform_err; + } + } + + if (codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open codec %s\n", + codec_dai->name); + goto codec_dai_err; + } + } + + if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + ret = rtd->dai_link->ops->startup(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); + goto machine_err; + } + } + + /* Check that the codec and cpu DAIs are compatible */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.rate_min = + max(codec_dai_drv->playback.rate_min, + cpu_dai_drv->playback.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->playback.rate_max, + cpu_dai_drv->playback.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->playback.channels_min, + cpu_dai_drv->playback.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->playback.channels_max, + cpu_dai_drv->playback.channels_max); + runtime->hw.formats = + codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; + runtime->hw.rates = + codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; + if (codec_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->playback.rates; + if (cpu_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->playback.rates; + } else { + runtime->hw.rate_min = + max(codec_dai_drv->capture.rate_min, + cpu_dai_drv->capture.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->capture.rate_max, + cpu_dai_drv->capture.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->capture.channels_min, + cpu_dai_drv->capture.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->capture.channels_max, + cpu_dai_drv->capture.channels_max); + runtime->hw.formats = + codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; + runtime->hw.rates = + codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; + if (codec_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->capture.rates; + if (cpu_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->capture.rates; + } + + ret = -EINVAL; + snd_pcm_limit_hw_rates(runtime); + if (!runtime->hw.rates) { + printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.formats) { + printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { + printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + + /* Symmetry only applies if we've already got an active stream. */ + if (cpu_dai->active || codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream); + if (ret != 0) + goto config_err; + } + + pr_debug("asoc: %s <-> %s info:\n", + codec_dai->name, cpu_dai->name); + pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); + pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, + runtime->hw.channels_max); + pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, + runtime->hw.rate_max); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + cpu_dai->active++; + codec_dai->active++; + rtd->codec->active++; + mutex_unlock(&pcm_mutex); + return 0; + +config_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + +machine_err: + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + +codec_dai_err: + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + +platform_err: + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); +out: + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Power down the audio subsystem pmdown_time msecs after close is called. + * This is to ensure there are no pops or clicks in between any music tracks + * due to DAPM power cycling. + */ +static void close_delayed_work(struct work_struct *work) +{ + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + mutex_lock(&pcm_mutex); + + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->driver->playback.stream_name, + codec_dai->playback_active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); + + /* are we waiting on this codec DAI stream */ + if (codec_dai->pop_wait == 1) { + codec_dai->pop_wait = 0; + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&pcm_mutex); +} + +/* + * Called by ALSA when a PCM substream is closed. Private data can be + * freed here. The cpu DAI, codec DAI, machine and platform are also + * shutdown. + */ +static int soc_codec_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock(&pcm_mutex); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + cpu_dai->active--; + codec_dai->active--; + codec->active--; + + /* Muting the DAC suppresses artifacts caused during digital + * shutdown, for example from stopping clocks. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_digital_mute(codec_dai, 1); + + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); + + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + cpu_dai->runtime = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* start delayed pop wq here for playback streams */ + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } else { + /* capture streams can be powered down now */ + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&pcm_mutex); + return 0; +} + +/* + * Called by ALSA when the PCM substream is prepared, can set format, sample + * rate, etc. This function is non atomic and can be called multiple times, + * it can refer to the runtime info. + */ +static int soc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock(&pcm_mutex); + + if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + ret = rtd->dai_link->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: machine prepare error\n"); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->prepare) { + ret = platform->driver->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: platform prepare error\n"); + goto out; + } + } + + if (codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: codec DAI prepare error\n"); + goto out; + } + } + + if (cpu_dai->driver->ops->prepare) { + ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: cpu DAI prepare error\n"); + goto out; + } + } + + /* cancel any delayed stream shutdown that is pending */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->pop_wait) { + codec_dai->pop_wait = 0; + cancel_delayed_work(&rtd->delayed_work); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_START); + + snd_soc_dai_digital_mute(codec_dai, 0); + +out: + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Called by ALSA when the hardware params are set by application. This + * function can also be called multiple times and can allocate buffers + * (using snd_pcm_lib_* ). It's non-atomic. + */ +static int soc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock(&pcm_mutex); + + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + ret = rtd->dai_link->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: machine hw_params failed\n"); + goto out; + } + } + + if (codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't set codec %s hw params\n", + codec_dai->name); + goto codec_err; + } + } + + if (cpu_dai->driver->ops->hw_params) { + ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: interface %s hw params failed\n", + cpu_dai->name); + goto interface_err; + } + } + + if (platform->driver->ops && platform->driver->ops->hw_params) { + ret = platform->driver->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: platform %s hw params failed\n", + platform->name); + goto platform_err; + } + } + + rtd->rate = params_rate(params); + +out: + mutex_unlock(&pcm_mutex); + return ret; + +platform_err: + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + +interface_err: + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + +codec_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Frees resources allocated by hw_params, can be called multiple times + */ +static int soc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock(&pcm_mutex); + + /* apply codec digital mute */ + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); + + /* free any machine hw params */ + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + /* free any DMA resources */ + if (platform->driver->ops && platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); + + /* now free hw params for the DAIs */ + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + + mutex_unlock(&pcm_mutex); + return 0; +} + +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->ops && platform->driver->ops->trigger) { + ret = platform->driver->ops->trigger(substream, cmd); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->trigger) { + ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} + +/* + * soc level wrapper for pointer callback + * If cpu_dai, codec_dai, platform driver has the delay callback, than + * the runtime->delay will be updated accordingly. + */ +static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t offset = 0; + snd_pcm_sframes_t delay = 0; + + if (platform->driver->ops && platform->driver->ops->pointer) + offset = platform->driver->ops->pointer(substream); + + if (cpu_dai->driver->ops->delay) + delay += cpu_dai->driver->ops->delay(substream, cpu_dai); + + if (codec_dai->driver->ops->delay) + delay += codec_dai->driver->ops->delay(substream, codec_dai); + + if (platform->driver->delay) + delay += platform->driver->delay(substream, codec_dai); + + runtime->delay = delay; + + return offset; +} + +/* ASoC PCM operations */ +static struct snd_pcm_ops soc_pcm_ops = { + .open = soc_pcm_open, + .close = soc_codec_close, + .hw_params = soc_pcm_hw_params, + .hw_free = soc_pcm_hw_free, + .prepare = soc_pcm_prepare, + .trigger = soc_pcm_trigger, + .pointer = soc_pcm_pointer, +}; + +/* create a new pcm */ +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm *pcm; + char new_name[64]; + int ret = 0, playback = 0, capture = 0; + + /* check client and interface hw capabilities */ + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + ret = snd_pcm_new(rtd->card->snd_card, new_name, + num, playback, capture, &pcm); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + return ret; + } + + /* DAPM dai link stream work */ + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + + rtd->pcm = pcm; + pcm->private_data = rtd; + if (platform->driver->ops) { + soc_pcm_ops.mmap = platform->driver->ops->mmap; + soc_pcm_ops.pointer = platform->driver->ops->pointer; + soc_pcm_ops.ioctl = platform->driver->ops->ioctl; + soc_pcm_ops.copy = platform->driver->ops->copy; + soc_pcm_ops.silence = platform->driver->ops->silence; + soc_pcm_ops.ack = platform->driver->ops->ack; + soc_pcm_ops.page = platform->driver->ops->page; + } + + if (playback) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + + if (capture) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + pr_err("asoc: platform pcm constructor failed\n"); + return ret; + } + } + + pcm->private_free = platform->driver->pcm_free; + printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + cpu_dai->name); + return ret; +} -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: snd_soc_new_{mixer, mux, pga} make sure to use right DAPM context
by Lars-Peter Clausen 09 Jun '11

09 Jun '11

Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a DAPM context not matching the widgets context. This can lead to a wrong prefix_len calculation, which will result in undefined behaviour. To avoid this always use the DAPM context from the widget itself. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org --- sound/soc/soc-dapm.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 34106bc..fd2d774 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -337,9 +337,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, } /* create new dapm mixer control */ -static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mixer(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; int i, ret = 0; size_t name_len, prefix_len; struct snd_soc_dapm_path *path; @@ -437,9 +437,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, } /* create new dapm mux control */ -static int dapm_new_mux(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mux(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; @@ -522,8 +522,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, } /* create new dapm volume control */ -static int dapm_new_pga(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_pga(struct snd_soc_dapm_widget *w) { if (w->num_kcontrols) dev_err(w->dapm->dev, @@ -1838,13 +1837,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: w->power_check = dapm_generic_check_power; - dapm_new_mixer(dapm, w); + dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; - dapm_new_mux(dapm, w); + dapm_new_mux(w); break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: @@ -1857,7 +1856,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: w->power_check = dapm_generic_check_power; - dapm_new_pga(dapm, w); + dapm_new_pga(w); break; case snd_soc_dapm_input: case snd_soc_dapm_output: -- 1.7.2.5

3 2

Re: [alsa-devel] [PATCH 0/1] Submitting MAX98088 codec patch
by Mark Brown 09 Jun '11

09 Jun '11

On Thu, Jun 09, 2011 at 05:50:16PM +0900, Jin Park wrote: > The max98088 codec driver had unnecessary duplicated codes on DAI operations. > For example, max98088_dai1_hw_params and max98088_dai2_hw_params, > they are almost same, except only register offsets. > So I have combined DAI operations for DAI1 and DAI2. > Please review my patch. Note that there's no need to have a cover letter for a single patch - any useful information should be in the patch itself.

2 1

[alsa-devel] [PATCH 1/2] ASoC: p1022ds: fix incorrect referencing of device tree properties
by Timur Tabi 09 Jun '11

09 Jun '11

Device tree integer properties are encoded in big-endian format, but some of the Freescale ASoC drivers were assuming that the host is in big-endian format as well. Although this is true, it's better to use endian-safe accessors. Also add a check for a failed ioremap() call in the SSI driver. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_ssi.c | 9 +++++++-- sound/soc/fsl/mpc8610_hpcd.c | 10 +++++----- sound/soc/fsl/p1022_ds.c | 10 +++++----- 4 files changed, 18 insertions(+), 13 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 50b5df8..3872598 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -940,7 +940,7 @@ static int __devinit fsl_soc_dma_probe(struct platform_device *pdev) iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); if (iprop) - dma->ssi_fifo_depth = *iprop; + dma->ssi_fifo_depth = be32_to_cpup(iprop); else /* Older 8610 DTs didn't have the fifo-depth property */ dma->ssi_fifo_depth = 8; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 313e0cc..d48afea 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -678,7 +678,12 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) kfree(ssi_private); return ret; } - ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start); + ssi_private->ssi = of_iomap(np, 0); + if (!ssi_private->ssi) { + dev_err(&pdev->dev, "could not map device resources\n"); + kfree(ssi_private); + return -ENOMEM; + } ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); @@ -691,7 +696,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Determine the FIFO depth. */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); if (iprop) - ssi_private->fifo_depth = *iprop; + ssi_private->fifo_depth = be32_to_cpup(iprop); else /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index c16c6b2..a192979 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -233,7 +233,7 @@ static int get_parent_cell_index(struct device_node *np) if (!iprop) return -1; - return *iprop; + return be32_to_cpup(iprop); } /** @@ -258,7 +258,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (!iprop) return -EINVAL; - addr = *iprop; + addr = be32_to_cpup(iprop); bus = get_parent_cell_index(np); if (bus < 0) @@ -305,7 +305,7 @@ static int get_dma_channel(struct device_node *ssi_np, return -EINVAL; } - *dma_channel_id = *iprop; + *dma_channel_id = be32_to_cpup(iprop); *dma_id = get_parent_cell_index(dma_channel_np); of_node_put(dma_channel_np); @@ -379,7 +379,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - machine_data->ssi_id = *iprop; + machine_data->ssi_id = be32_to_cpup(iprop); /* Get the serial format and clock direction. */ sprop = of_get_property(np, "fsl,mode", NULL); @@ -405,7 +405,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - machine_data->clk_frequency = *iprop; + machine_data->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { machine_data->dai_format = SND_SOC_DAIFMT_I2S; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 66e0b68..8fa4d5f 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -232,7 +232,7 @@ static int get_parent_cell_index(struct device_node *np) iprop = of_get_property(parent, "cell-index", NULL); if (iprop) - ret = *iprop; + ret = be32_to_cpup(iprop); of_node_put(parent); @@ -261,7 +261,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (!iprop) return -EINVAL; - addr = *iprop; + addr = be32_to_cpup(iprop); bus = get_parent_cell_index(np); if (bus < 0) @@ -308,7 +308,7 @@ static int get_dma_channel(struct device_node *ssi_np, return -EINVAL; } - *dma_channel_id = *iprop; + *dma_channel_id = be32_to_cpup(iprop); *dma_id = get_parent_cell_index(dma_channel_np); of_node_put(dma_channel_np); @@ -379,7 +379,7 @@ static int p1022_ds_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - mdata->ssi_id = *iprop; + mdata->ssi_id = be32_to_cpup(iprop); /* Get the serial format and clock direction. */ sprop = of_get_property(np, "fsl,mode", NULL); @@ -405,7 +405,7 @@ static int p1022_ds_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - mdata->clk_frequency = *iprop; + mdata->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { mdata->dai_format = SND_SOC_DAIFMT_I2S; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; -- 1.7.3.4

5 7

[alsa-devel] [PATCH] ASoC: core - Separate out PCM operations into new file.
by Liam Girdwood 09 Jun '11

09 Jun '11

In preparation for Dynamic PCM support (AKA DSP support). There will be future patches that add support to allow PCMs to be dynamically routed to multiple DAIs at startup and also during stream runtime. This patch moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will in simplify the ASoC core features into distinct files. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 612 +----------------------------------------------- sound/soc/soc-pcm.c | 645 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 647 insertions(+), 612 deletions(-) create mode 100644 sound/soc/soc-pcm.c diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1ed61c5..adb5719 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o soc-pcm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index df5608d..32d7d2f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -44,7 +44,6 @@ #define NAME_SIZE 32 -static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS @@ -58,7 +57,7 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -485,552 +484,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && - !rtd->dai_link->symmetric_rates) - return 0; - - /* This can happen if multiple streams are starting simultaneously - - * the second can need to get its constraints before the first has - * picked a rate. Complain and allow the application to carry on. - */ - if (!rtd->rate) { - dev_warn(&rtd->dev, - "Not enforcing symmetric_rates due to race\n"); - return 0; - } - - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } - - return 0; -} - -/* - * Called by ALSA when a PCM substream is opened, the runtime->hw record is - * then initialized and any private data can be allocated. This also calls - * startup for the cpu DAI, platform, machine and codec DAI. - */ -static int soc_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; - - mutex_lock(&pcm_mutex); - - /* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { - ret = cpu_dai->driver->ops->startup(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->open) { - ret = platform->driver->ops->open(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); - goto platform_err; - } - } - - if (codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); - goto codec_dai_err; - } - } - - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { - ret = rtd->dai_link->ops->startup(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); - goto machine_err; - } - } - - /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = - max(codec_dai_drv->playback.rate_min, - cpu_dai_drv->playback.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->playback.rate_max, - cpu_dai_drv->playback.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->playback.channels_min, - cpu_dai_drv->playback.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->playback.channels_max, - cpu_dai_drv->playback.channels_max); - runtime->hw.formats = - codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; - runtime->hw.rates = - codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; - if (codec_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->playback.rates; - if (cpu_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->playback.rates; - } else { - runtime->hw.rate_min = - max(codec_dai_drv->capture.rate_min, - cpu_dai_drv->capture.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->capture.rate_max, - cpu_dai_drv->capture.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->capture.channels_min, - cpu_dai_drv->capture.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->capture.channels_max, - cpu_dai_drv->capture.channels_max); - runtime->hw.formats = - codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; - runtime->hw.rates = - codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; - if (codec_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->capture.rates; - if (cpu_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->capture.rates; - } - - ret = -EINVAL; - snd_pcm_limit_hw_rates(runtime); - if (!runtime->hw.rates) { - printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.formats) { - printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.channels_min || !runtime->hw.channels_max || - runtime->hw.channels_min > runtime->hw.channels_max) { - printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - - /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); - if (ret != 0) - goto config_err; - } - - pr_debug("asoc: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); - pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); - pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, - runtime->hw.channels_max); - pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, - runtime->hw.rate_max); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active++; - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - codec_dai->capture_active++; - } - cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; - mutex_unlock(&pcm_mutex); - return 0; - -config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - -machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - -codec_dai_err: - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - -platform_err: - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Power down the audio subsystem pmdown_time msecs after close is called. - * This is to ensure there are no pops or clicks in between any music tracks - * due to DAPM power cycling. - */ -static void close_delayed_work(struct work_struct *work) -{ - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - mutex_lock(&pcm_mutex); - - pr_debug("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->driver->playback.stream_name, - codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); - - /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); -} - -/* - * Called by ALSA when a PCM substream is closed. Private data can be - * freed here. The cpu DAI, codec DAI, machine and platform are also - * shutdown. - */ -static int soc_codec_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active--; - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - codec_dai->capture_active--; - } - - cpu_dai->active--; - codec_dai->active--; - codec->active--; - - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); - - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); - - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - cpu_dai->runtime = NULL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); - } else { - /* capture streams can be powered down now */ - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); - return 0; -} - -/* - * Called by ALSA when the PCM substream is prepared, can set format, sample - * rate, etc. This function is non atomic and can be called multiple times, - * it can refer to the runtime info. - */ -static int soc_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { - ret = rtd->dai_link->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: machine prepare error\n"); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->prepare) { - ret = platform->driver->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: platform prepare error\n"); - goto out; - } - } - - if (codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: codec DAI prepare error\n"); - goto out; - } - } - - if (cpu_dai->driver->ops->prepare) { - ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: cpu DAI prepare error\n"); - goto out; - } - } - - /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; - cancel_delayed_work(&rtd->delayed_work); - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_START); - - snd_soc_dai_digital_mute(codec_dai, 0); - -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Called by ALSA when the hardware params are set by application. This - * function can also be called multiple times and can allocate buffers - * (using snd_pcm_lib_* ). It's non-atomic. - */ -static int soc_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { - ret = rtd->dai_link->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: machine hw_params failed\n"); - goto out; - } - } - - if (codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); - goto codec_err; - } - } - - if (cpu_dai->driver->ops->hw_params) { - ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); - goto interface_err; - } - } - - if (platform->driver->ops && platform->driver->ops->hw_params) { - ret = platform->driver->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); - goto platform_err; - } - } - - rtd->rate = params_rate(params); - -out: - mutex_unlock(&pcm_mutex); - return ret; - -platform_err: - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - -interface_err: - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - -codec_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Frees resources allocated by hw_params, can be called multiple times - */ -static int soc_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - /* apply codec digital mute */ - if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); - - /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - /* free any DMA resources */ - if (platform->driver->ops && platform->driver->ops->hw_free) - platform->driver->ops->hw_free(substream); - - /* now free hw params for the DAIs */ - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - - mutex_unlock(&pcm_mutex); - return 0; -} - -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; - } - - if (platform->driver->ops && platform->driver->ops->trigger) { - ret = platform->driver->ops->trigger(substream, cmd); - if (ret < 0) - return ret; - } - - if (cpu_dai->driver->ops->trigger) { - ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; - } - return 0; -} - -/* - * soc level wrapper for pointer callback - * If cpu_dai, codec_dai, platform driver has the delay callback, than - * the runtime->delay will be updated accordingly. - */ -static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t offset = 0; - snd_pcm_sframes_t delay = 0; - - if (platform->driver->ops && platform->driver->ops->pointer) - offset = platform->driver->ops->pointer(substream); - - if (cpu_dai->driver->ops->delay) - delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - - if (codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); - - if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); - - runtime->delay = delay; - - return offset; -} - -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_codec_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - #ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ int snd_soc_suspend(struct device *dev) @@ -1692,9 +1145,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (order != SND_SOC_COMP_ORDER_LAST) return 0; - /* DAPM dai link stream work */ - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); - ret = soc_post_component_init(card, codec, num, 0); if (ret) return ret; @@ -2114,66 +1564,6 @@ static struct platform_driver soc_driver = { .remove = soc_remove, }; -/* create a new pcm */ -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_pcm *pcm; - char new_name[64]; - int ret = 0, playback = 0, capture = 0; - - /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - - if (codec_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min) - capture = 1; - - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); - ret = snd_pcm_new(rtd->card->snd_card, new_name, - num, playback, capture, &pcm); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); - return ret; - } - - rtd->pcm = pcm; - pcm->private_data = rtd; - if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; - } - - if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); - - if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - - if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd); - if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); - return ret; - } - } - - pcm->private_free = platform->driver->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); - return ret; -} - /** * snd_soc_codec_volatile_register: Report if a register is volatile. * diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c new file mode 100644 index 0000000..7b725be --- /dev/null +++ b/sound/soc/soc-pcm.c @@ -0,0 +1,645 @@ +/* + * soc-pcm.c -- ALSA SoC PCM + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Authors: Liam Girdwood <lrg(a)ti.com> + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/debugfs.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/ac97_codec.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> + +static DEFINE_MUTEX(pcm_mutex); + +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (!codec_dai->driver->symmetric_rates && + !cpu_dai->driver->symmetric_rates && + !rtd->dai_link->symmetric_rates) + return 0; + + /* This can happen if multiple streams are starting simultaneously - + * the second can need to get its constraints before the first has + * picked a rate. Complain and allow the application to carry on. + */ + if (!rtd->rate) { + dev_warn(&rtd->dev, + "Not enforcing symmetric_rates due to race\n"); + return 0; + } + + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + rtd->rate, rtd->rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + + return 0; +} + +/* + * Called by ALSA when a PCM substream is opened, the runtime->hw record is + * then initialized and any private data can be allocated. This also calls + * startup for the cpu DAI, platform, machine and codec DAI. + */ +static int soc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; + int ret = 0; + + mutex_lock(&pcm_mutex); + + /* startup the audio subsystem */ + if (cpu_dai->driver->ops->startup) { + ret = cpu_dai->driver->ops->startup(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open interface %s\n", + cpu_dai->name); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->open) { + ret = platform->driver->ops->open(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + goto platform_err; + } + } + + if (codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open codec %s\n", + codec_dai->name); + goto codec_dai_err; + } + } + + if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + ret = rtd->dai_link->ops->startup(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); + goto machine_err; + } + } + + /* Check that the codec and cpu DAIs are compatible */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.rate_min = + max(codec_dai_drv->playback.rate_min, + cpu_dai_drv->playback.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->playback.rate_max, + cpu_dai_drv->playback.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->playback.channels_min, + cpu_dai_drv->playback.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->playback.channels_max, + cpu_dai_drv->playback.channels_max); + runtime->hw.formats = + codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; + runtime->hw.rates = + codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; + if (codec_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->playback.rates; + if (cpu_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->playback.rates; + } else { + runtime->hw.rate_min = + max(codec_dai_drv->capture.rate_min, + cpu_dai_drv->capture.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->capture.rate_max, + cpu_dai_drv->capture.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->capture.channels_min, + cpu_dai_drv->capture.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->capture.channels_max, + cpu_dai_drv->capture.channels_max); + runtime->hw.formats = + codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; + runtime->hw.rates = + codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; + if (codec_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->capture.rates; + if (cpu_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->capture.rates; + } + + ret = -EINVAL; + snd_pcm_limit_hw_rates(runtime); + if (!runtime->hw.rates) { + printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.formats) { + printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { + printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + + /* Symmetry only applies if we've already got an active stream. */ + if (cpu_dai->active || codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream); + if (ret != 0) + goto config_err; + } + + pr_debug("asoc: %s <-> %s info:\n", + codec_dai->name, cpu_dai->name); + pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); + pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, + runtime->hw.channels_max); + pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, + runtime->hw.rate_max); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + cpu_dai->active++; + codec_dai->active++; + rtd->codec->active++; + mutex_unlock(&pcm_mutex); + return 0; + +config_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + +machine_err: + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + +codec_dai_err: + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + +platform_err: + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); +out: + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Power down the audio subsystem pmdown_time msecs after close is called. + * This is to ensure there are no pops or clicks in between any music tracks + * due to DAPM power cycling. + */ +static void close_delayed_work(struct work_struct *work) +{ + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + mutex_lock(&pcm_mutex); + + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->driver->playback.stream_name, + codec_dai->playback_active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); + + /* are we waiting on this codec DAI stream */ + if (codec_dai->pop_wait == 1) { + codec_dai->pop_wait = 0; + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&pcm_mutex); +} + +/* + * Called by ALSA when a PCM substream is closed. Private data can be + * freed here. The cpu DAI, codec DAI, machine and platform are also + * shutdown. + */ +static int soc_codec_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock(&pcm_mutex); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + cpu_dai->active--; + codec_dai->active--; + codec->active--; + + /* Muting the DAC suppresses artifacts caused during digital + * shutdown, for example from stopping clocks. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_digital_mute(codec_dai, 1); + + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); + + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + cpu_dai->runtime = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* start delayed pop wq here for playback streams */ + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } else { + /* capture streams can be powered down now */ + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&pcm_mutex); + return 0; +} + +/* + * Called by ALSA when the PCM substream is prepared, can set format, sample + * rate, etc. This function is non atomic and can be called multiple times, + * it can refer to the runtime info. + */ +static int soc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock(&pcm_mutex); + + if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + ret = rtd->dai_link->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: machine prepare error\n"); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->prepare) { + ret = platform->driver->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: platform prepare error\n"); + goto out; + } + } + + if (codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: codec DAI prepare error\n"); + goto out; + } + } + + if (cpu_dai->driver->ops->prepare) { + ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: cpu DAI prepare error\n"); + goto out; + } + } + + /* cancel any delayed stream shutdown that is pending */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->pop_wait) { + codec_dai->pop_wait = 0; + cancel_delayed_work(&rtd->delayed_work); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_START); + + snd_soc_dai_digital_mute(codec_dai, 0); + +out: + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Called by ALSA when the hardware params are set by application. This + * function can also be called multiple times and can allocate buffers + * (using snd_pcm_lib_* ). It's non-atomic. + */ +static int soc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock(&pcm_mutex); + + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + ret = rtd->dai_link->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: machine hw_params failed\n"); + goto out; + } + } + + if (codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't set codec %s hw params\n", + codec_dai->name); + goto codec_err; + } + } + + if (cpu_dai->driver->ops->hw_params) { + ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: interface %s hw params failed\n", + cpu_dai->name); + goto interface_err; + } + } + + if (platform->driver->ops && platform->driver->ops->hw_params) { + ret = platform->driver->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: platform %s hw params failed\n", + platform->name); + goto platform_err; + } + } + + rtd->rate = params_rate(params); + +out: + mutex_unlock(&pcm_mutex); + return ret; + +platform_err: + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + +interface_err: + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + +codec_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Frees resources allocated by hw_params, can be called multiple times + */ +static int soc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock(&pcm_mutex); + + /* apply codec digital mute */ + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); + + /* free any machine hw params */ + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + /* free any DMA resources */ + if (platform->driver->ops && platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); + + /* now free hw params for the DAIs */ + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + + mutex_unlock(&pcm_mutex); + return 0; +} + +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->ops && platform->driver->ops->trigger) { + ret = platform->driver->ops->trigger(substream, cmd); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->trigger) { + ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} + +/* + * soc level wrapper for pointer callback + * If cpu_dai, codec_dai, platform driver has the delay callback, than + * the runtime->delay will be updated accordingly. + */ +static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t offset = 0; + snd_pcm_sframes_t delay = 0; + + if (platform->driver->ops && platform->driver->ops->pointer) + offset = platform->driver->ops->pointer(substream); + + if (cpu_dai->driver->ops->delay) + delay += cpu_dai->driver->ops->delay(substream, cpu_dai); + + if (codec_dai->driver->ops->delay) + delay += codec_dai->driver->ops->delay(substream, codec_dai); + + if (platform->driver->delay) + delay += platform->driver->delay(substream, codec_dai); + + runtime->delay = delay; + + return offset; +} + +/* ASoC PCM operations */ +static struct snd_pcm_ops soc_pcm_ops = { + .open = soc_pcm_open, + .close = soc_codec_close, + .hw_params = soc_pcm_hw_params, + .hw_free = soc_pcm_hw_free, + .prepare = soc_pcm_prepare, + .trigger = soc_pcm_trigger, + .pointer = soc_pcm_pointer, +}; + +/* create a new pcm */ +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm *pcm; + char new_name[64]; + int ret = 0, playback = 0, capture = 0; + + /* check client and interface hw capabilities */ + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + ret = snd_pcm_new(rtd->card->snd_card, new_name, + num, playback, capture, &pcm); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + return ret; + } + + /* DAPM dai link stream work */ + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + + rtd->pcm = pcm; + pcm->private_data = rtd; + if (platform->driver->ops) { + soc_pcm_ops.mmap = platform->driver->ops->mmap; + soc_pcm_ops.pointer = platform->driver->ops->pointer; + soc_pcm_ops.ioctl = platform->driver->ops->ioctl; + soc_pcm_ops.copy = platform->driver->ops->copy; + soc_pcm_ops.silence = platform->driver->ops->silence; + soc_pcm_ops.ack = platform->driver->ops->ack; + soc_pcm_ops.page = platform->driver->ops->page; + } + + if (playback) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + + if (capture) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + pr_err("asoc: platform pcm constructor failed\n"); + return ret; + } + } + + pcm->private_free = platform->driver->pcm_free; + printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + cpu_dai->name); + return ret; +} -- 1.7.4.1

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Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 09 Jun '11

09 Jun '11

At Thu, 9 Jun 2011 11:28:26 +0200, Michal Hocko wrote: > > [1 <text/plain; us-ascii (7bit)>] > On Thu 09-06-11 11:20:04, Takashi Iwai wrote: > > At Thu, 9 Jun 2011 11:06:07 +0200, > > Michal Hocko wrote: > > > > > > On Thu 09-06-11 10:56:16, Michal Hocko wrote: > > > > On Thu 09-06-11 08:09:18, Takashi Iwai wrote: > > > > > At Wed, 8 Jun 2011 19:48:55 +0200, > > > [...] > > > > > Thanks, that's helpful. > > > > > As a quirk solution, could you try to pass model=auto option to > > > > > snd-hda-intel module? > > > > > > > > Headphones work now but speakers are quite... I have tried to play > > > > with Headphone and Speaker controls in alsamixer but no changes. > > > > > > > > Here is what I get in logs when the module is loaded: > > > > [ 5320.360277] HDA Intel 0000:00:1b.0: PCI INT A disabled > > > > [ 5326.920247] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > > > [ 5326.920262] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > > > [ 5326.920279] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 > > > > [ 5326.920375] HDA Intel 0000:00:1b.0: irq 42 for MSI/MSI-X > > > > [ 5326.920405] HDA Intel 0000:00:1b.0: setting latency timer to 64 > > > > [ 5326.981085] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input18 > > > > [ 5326.981262] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input19 > > > > [ 5326.987934] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input20 > > > > > > > > [maybe not important but input numbers are increasing for every new load] > > > > > > > > There is another strange thing. After I stop mplayer and wait for some > > > > time (~1min) I can hear a "click" sound and then there is no sound at > > > > all... If I start mplayer in the meantime it just works... > > > > If I try to remove the module and load it again then things work and stop > > > > working after first playback again. > > > > > > And I have just found out that I do not have to start any playback to > > > see this. It is sufficient to load the module and wait a minute and > > > there is no sound at all (both from speakers or headphones). > > > > OK, could you give alsa-info.sh outputs at both working and > > non-working states? > > Sure. Attached. The only change seems to be: > Node 0x02 [Audio Output] wcaps 0x11: Stereo > Device: name="ALC262 Analog", type="Audio", device=0 > - Converter: stream=5, channel=0 > + Converter: stream=0, channel=0 This is OK. It shows that the converter was assigned once to this stream when you played back a sound before power-save. As long as the codec registers show the same value, it's not obviously what went wrong. The symptom sounds like a resume issue. If so, S3/S4 might show the same problem. BTW, does my patch fix the problem when you load without model=auto? If yes, does the problem appear with it? I.e. is the silence problem specific to model=auto or not? thanks, Takashi

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Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 09 Jun '11

09 Jun '11

At Thu, 9 Jun 2011 11:06:07 +0200, Michal Hocko wrote: > > On Thu 09-06-11 10:56:16, Michal Hocko wrote: > > On Thu 09-06-11 08:09:18, Takashi Iwai wrote: > > > At Wed, 8 Jun 2011 19:48:55 +0200, > [...] > > > Thanks, that's helpful. > > > As a quirk solution, could you try to pass model=auto option to > > > snd-hda-intel module? > > > > Headphones work now but speakers are quite... I have tried to play > > with Headphone and Speaker controls in alsamixer but no changes. > > > > Here is what I get in logs when the module is loaded: > > [ 5320.360277] HDA Intel 0000:00:1b.0: PCI INT A disabled > > [ 5326.920247] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > [ 5326.920262] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > [ 5326.920279] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 > > [ 5326.920375] HDA Intel 0000:00:1b.0: irq 42 for MSI/MSI-X > > [ 5326.920405] HDA Intel 0000:00:1b.0: setting latency timer to 64 > > [ 5326.981085] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input18 > > [ 5326.981262] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input19 > > [ 5326.987934] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input20 > > > > [maybe not important but input numbers are increasing for every new load] > > > > There is another strange thing. After I stop mplayer and wait for some > > time (~1min) I can hear a "click" sound and then there is no sound at > > all... If I start mplayer in the meantime it just works... > > If I try to remove the module and load it again then things work and stop > > working after first playback again. > > And I have just found out that I do not have to start any playback to > see this. It is sufficient to load the module and wait a minute and > there is no sound at all (both from speakers or headphones). OK, could you give alsa-info.sh outputs at both working and non-working states? thanks, Takashi

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Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 09 Jun '11

09 Jun '11

At Thu, 9 Jun 2011 10:56:16 +0200, Michal Hocko wrote: > > On Thu 09-06-11 08:09:18, Takashi Iwai wrote: > > At Wed, 8 Jun 2011 19:48:55 +0200, > > Michal Hocko wrote: > > > > > > On Wed 08-06-11 18:03:24, Takashi Iwai wrote: > > > > At Tue, 7 Jun 2011 20:05:58 +0200, > > > > Michal Hocko wrote: > > > > > > > > > > Hi, > > > > > before 3.0.0-rc1, when I plugged anything into headphone output of my > > > > > laptop, speakers were disabled and the sound has been routed to the > > > > > headphone output. This doesn't seem to work anymore starting with > > > > > 3.0.0-rc1. Speakers stay on even if I plug something to the headphone > > > > > output. > > > > > Is this a bug or a new behavior that userspace has to cope > > > > > with? > > > > > > > > > > I am sorry for the lack of information but I have no idea what to > > > > > provide. > > > > > > > > Please give alsa-info.sh output at best. Run it with --no-upload > > > > option, and attach the output file. This will contain most of > > > > necessary information. > > > > > > See attached. I have added also .39 output where things work as expected > > > (just in case it could be useful). > > > > Thanks, that's helpful. > > As a quirk solution, could you try to pass model=auto option to > > snd-hda-intel module? > > Headphones work now but speakers are quite... I have tried to play > with Headphone and Speaker controls in alsamixer but no changes. > > Here is what I get in logs when the module is loaded: > [ 5320.360277] HDA Intel 0000:00:1b.0: PCI INT A disabled > [ 5326.920247] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > [ 5326.920262] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > [ 5326.920279] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 > [ 5326.920375] HDA Intel 0000:00:1b.0: irq 42 for MSI/MSI-X > [ 5326.920405] HDA Intel 0000:00:1b.0: setting latency timer to 64 > [ 5326.981085] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input18 > [ 5326.981262] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input19 > [ 5326.987934] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input20 > > [maybe not important but input numbers are increasing for every new load] No bug, it's the implementation issue of input layer :) > There is another strange thing. After I stop mplayer and wait for some > time (~1min) I can hear a "click" sound and then there is no sound at > all... If I start mplayer in the meantime it just works... This is possibly the powersave. It means that the resume doesn't work properly after powersave. > If I try to remove the module and load it again then things work and stop > working after first playback again. > > > You may need to adjust the mixer after changing > > this, but it should provide the full (even better) functionality. > > It'll support also the auto-mic jack switching. > > $ amixer | grep -i "auto.*mic" > What is the control to look at? There is no control. Simply plugging the mic jack will switch the recording source automatically. Takashi

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[alsa-devel] Retrieving the number of input channels on a sound card
by André Prado 09 Jun '11

09 Jun '11

Hello. I am new to Alsa programming and i want to get the number of input channels on an audio device. I am reading a lot of tutorials and documentations but i am struggling to do this. Here is a code i've saw to Lists the hardware names of MIDI output device/sub-devices. I've understood it. But i want to get the number of input channels. Any help is appreciated. THanks // listmidi.c // Lists the hardware names of MIDI output device/sub-devices // upon each ALSA sound card in the system. // // Compile as: // gcc -o listmidi listmidi.c -lasound #include <stdio.h> #include <string.h> #include <alsa/asoundlib.h> int main(int argc, char **argv) { register int err; int cardNum; // Start with first card cardNum = -1; for (;;) { snd_ctl_t *cardHandle; // Get next sound card's card number. When "cardNum" == -1, then ALSA // fetches the first card if ((err = snd_card_next(&cardNum)) < 0) { printf("Can't get the next card number: %s\n", snd_strerror(err)); break; } // No more cards? ALSA sets "cardNum" to -1 if so if (cardNum < 0) break; // Open this card's control interface. We specify only the card number -- not // any device nor sub-device too { char str[64]; sprintf(str, "hw:%i", cardNum); if ((err = snd_ctl_open(&cardHandle, str, 0)) < 0) { printf("Can't open card %i: %s\n", cardNum, snd_strerror(err)); continue; } } { int devNum; // Start with the first MIDI device on this card devNum = -1; for (;;) { snd_rawmidi_info_t *rawMidiInfo; register int subDevCount, i; // Get the number of the next MIDI device on this card if ((err = snd_ctl_rawmidi_next_device(cardHandle, &devNum)) < 0) { printf("Can't get next MIDI device number: %s\n", snd_strerror(err)); break; } // No more MIDI devices on this card? ALSA sets "devNum" to -1 if so. // NOTE: It's possible that this sound card may have no MIDI devices on it // at all, for example if it's only a digital audio card if (devNum < 0) break; // To get some info about the subdevices of this MIDI device (on the card), we need a // snd_rawmidi_info_t, so let's allocate one on the stack snd_rawmidi_info_alloca(&rawMidiInfo); memset(rawMidiInfo, 0, snd_rawmidi_info_sizeof()); // Tell ALSA which device (number) we want info about snd_rawmidi_info_set_device(rawMidiInfo, devNum); // Get info on the MIDI outs of this device snd_rawmidi_info_set_stream(rawMidiInfo, SND_RAWMIDI_STREAM_OUTPUT); i = -1; subDevCount = 1; // More subdevices? while (++i < subDevCount) { // Tell ALSA to fill in our snd_rawmidi_info_t with info on this subdevice snd_rawmidi_info_set_subdevice(rawMidiInfo, i); if ((err = snd_ctl_rawmidi_info(cardHandle, rawMidiInfo)) < 0) { printf("Can't get info for MIDI output subdevice hw:%i,%i,%i: %s\n", cardNum, devNum, i, snd_strerror(err)); continue; } // Print out how many subdevices (once only) if (!i) { subDevCount = snd_rawmidi_info_get_subdevices_count(rawMidiInfo); printf("\nFound %i MIDI output subdevices on card %i\n", subDevCount, cardNum); } // NOTE: If there's only one subdevice, then the subdevice number is immaterial, // and can be omitted when you specify the hardware name printf((subDevCount > 1 ? " hw:%i,%i,%i\n" : " hw:%i,%i\n"), cardNum, devNum, i); } } } // Close the card's control interface after we're done with it snd_ctl_close(cardHandle); } snd_config_update_free_global(); } -- Att André

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Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 09 Jun '11

09 Jun '11

At Wed, 8 Jun 2011 19:48:55 +0200, Michal Hocko wrote: > > On Wed 08-06-11 18:03:24, Takashi Iwai wrote: > > At Tue, 7 Jun 2011 20:05:58 +0200, > > Michal Hocko wrote: > > > > > > Hi, > > > before 3.0.0-rc1, when I plugged anything into headphone output of my > > > laptop, speakers were disabled and the sound has been routed to the > > > headphone output. This doesn't seem to work anymore starting with > > > 3.0.0-rc1. Speakers stay on even if I plug something to the headphone > > > output. > > > Is this a bug or a new behavior that userspace has to cope > > > with? > > > > > > I am sorry for the lack of information but I have no idea what to > > > provide. > > > > Please give alsa-info.sh output at best. Run it with --no-upload > > option, and attach the output file. This will contain most of > > necessary information. > > See attached. I have added also .39 output where things work as expected > (just in case it could be useful). Thanks, that's helpful. As a quirk solution, could you try to pass model=auto option to snd-hda-intel module? You may need to adjust the mixer after changing this, but it should provide the full (even better) functionality. It'll support also the auto-mic jack switching. If auto-parser works with this hardware, we don't need to debug the old code but just switch to the auto-parser statically in the driver. thanks, Takashi

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