| /* |
| * soc-core.c -- ALSA SoC Audio Layer |
| * |
| * Copyright 2005 Wolfson Microelectronics PLC. |
| * Copyright 2005 Openedhand Ltd. |
| * |
| * Author: Liam Girdwood <lrg@slimlogic.co.uk> |
| * with code, comments and ideas from :- |
| * Richard Purdie <richard@openedhand.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. |
| * |
| * TODO: |
| * o Add hw rules to enforce rates, etc. |
| * o More testing with other codecs/machines. |
| * o Add more codecs and platforms to ensure good API coverage. |
| * o Support TDM on PCM and I2S |
| */ |
| |
| #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 <sound/core.h> |
| #include <sound/pcm.h> |
| #include <sound/pcm_params.h> |
| #include <sound/soc.h> |
| #include <sound/soc-dapm.h> |
| #include <sound/initval.h> |
| |
| static DEFINE_MUTEX(pcm_mutex); |
| static DEFINE_MUTEX(io_mutex); |
| static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); |
| |
| /* |
| * This is a timeout to do a DAPM powerdown after a stream is closed(). |
| * It can be used to eliminate pops between different playback streams, e.g. |
| * between two audio tracks. |
| */ |
| static int pmdown_time = 5000; |
| module_param(pmdown_time, int, 0); |
| MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); |
| |
| /* |
| * This function forces any delayed work to be queued and run. |
| */ |
| static int run_delayed_work(struct delayed_work *dwork) |
| { |
| int ret; |
| |
| /* cancel any work waiting to be queued. */ |
| ret = cancel_delayed_work(dwork); |
| |
| /* if there was any work waiting then we run it now and |
| * wait for it's completion */ |
| if (ret) { |
| schedule_delayed_work(dwork, 0); |
| flush_scheduled_work(); |
| } |
| return ret; |
| } |
| |
| #ifdef CONFIG_SND_SOC_AC97_BUS |
| /* unregister ac97 codec */ |
| static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) |
| { |
| if (codec->ac97->dev.bus) |
| device_unregister(&codec->ac97->dev); |
| return 0; |
| } |
| |
| /* stop no dev release warning */ |
| static void soc_ac97_device_release(struct device *dev){} |
| |
| /* register ac97 codec to bus */ |
| static int soc_ac97_dev_register(struct snd_soc_codec *codec) |
| { |
| int err; |
| |
| codec->ac97->dev.bus = &ac97_bus_type; |
| codec->ac97->dev.parent = NULL; |
| codec->ac97->dev.release = soc_ac97_device_release; |
| |
| dev_set_name(&codec->ac97->dev, "%d-%d:%s", |
| codec->card->number, 0, codec->name); |
| err = device_register(&codec->ac97->dev); |
| if (err < 0) { |
| snd_printk(KERN_ERR "Can't register ac97 bus\n"); |
| codec->ac97->dev.bus = NULL; |
| return err; |
| } |
| return 0; |
| } |
| #endif |
| |
| /* |
| * 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_soc_device *socdev = rtd->socdev; |
| struct snd_pcm_runtime *runtime = substream->runtime; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| int ret = 0; |
| |
| mutex_lock(&pcm_mutex); |
| |
| /* startup the audio subsystem */ |
| if (cpu_dai->ops.startup) { |
| ret = cpu_dai->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->pcm_ops->open) { |
| ret = platform->pcm_ops->open(substream); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); |
| goto platform_err; |
| } |
| } |
| |
| if (codec_dai->ops.startup) { |
| ret = codec_dai->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 (machine->ops && machine->ops->startup) { |
| ret = machine->ops->startup(substream); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: %s startup failed\n", machine->name); |
| goto machine_err; |
| } |
| } |
| |
| /* Check that the codec and cpu DAI's are compatible */ |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { |
| runtime->hw.rate_min = |
| max(codec_dai->playback.rate_min, |
| cpu_dai->playback.rate_min); |
| runtime->hw.rate_max = |
| min(codec_dai->playback.rate_max, |
| cpu_dai->playback.rate_max); |
| runtime->hw.channels_min = |
| max(codec_dai->playback.channels_min, |
| cpu_dai->playback.channels_min); |
| runtime->hw.channels_max = |
| min(codec_dai->playback.channels_max, |
| cpu_dai->playback.channels_max); |
| runtime->hw.formats = |
| codec_dai->playback.formats & cpu_dai->playback.formats; |
| runtime->hw.rates = |
| codec_dai->playback.rates & cpu_dai->playback.rates; |
| } else { |
| runtime->hw.rate_min = |
| max(codec_dai->capture.rate_min, |
| cpu_dai->capture.rate_min); |
| runtime->hw.rate_max = |
| min(codec_dai->capture.rate_max, |
| cpu_dai->capture.rate_max); |
| runtime->hw.channels_min = |
| max(codec_dai->capture.channels_min, |
| cpu_dai->capture.channels_min); |
| runtime->hw.channels_max = |
| min(codec_dai->capture.channels_max, |
| cpu_dai->capture.channels_max); |
| runtime->hw.formats = |
| codec_dai->capture.formats & cpu_dai->capture.formats; |
| runtime->hw.rates = |
| codec_dai->capture.rates & cpu_dai->capture.rates; |
| } |
| |
| 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 machine_err; |
| } |
| if (!runtime->hw.formats) { |
| printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", |
| codec_dai->name, cpu_dai->name); |
| goto machine_err; |
| } |
| if (!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 machine_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 = 1; |
| else |
| cpu_dai->capture.active = codec_dai->capture.active = 1; |
| cpu_dai->active = codec_dai->active = 1; |
| cpu_dai->runtime = runtime; |
| socdev->codec->active++; |
| mutex_unlock(&pcm_mutex); |
| return 0; |
| |
| machine_err: |
| if (machine->ops && machine->ops->shutdown) |
| machine->ops->shutdown(substream); |
| |
| codec_dai_err: |
| if (platform->pcm_ops->close) |
| platform->pcm_ops->close(substream); |
| |
| platform_err: |
| if (cpu_dai->ops.shutdown) |
| cpu_dai->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_device *socdev = |
| container_of(work, struct snd_soc_device, delayed_work.work); |
| struct snd_soc_codec *codec = socdev->codec; |
| struct snd_soc_dai *codec_dai; |
| int i; |
| |
| mutex_lock(&pcm_mutex); |
| for (i = 0; i < codec->num_dai; i++) { |
| codec_dai = &codec->dai[i]; |
| |
| pr_debug("pop wq checking: %s status: %s waiting: %s\n", |
| codec_dai->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) { |
| |
| /* Reduce power if no longer active */ |
| if (codec->active == 0) { |
| pr_debug("pop wq D1 %s %s\n", codec->name, |
| codec_dai->playback.stream_name); |
| snd_soc_dapm_set_bias_level(socdev, |
| SND_SOC_BIAS_PREPARE); |
| } |
| |
| codec_dai->pop_wait = 0; |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->playback.stream_name, |
| SND_SOC_DAPM_STREAM_STOP); |
| |
| /* Fall into standby if no longer active */ |
| if (codec->active == 0) { |
| pr_debug("pop wq D3 %s %s\n", codec->name, |
| codec_dai->playback.stream_name); |
| snd_soc_dapm_set_bias_level(socdev, |
| SND_SOC_BIAS_STANDBY); |
| } |
| } |
| } |
| 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_device *socdev = rtd->socdev; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| struct snd_soc_codec *codec = socdev->codec; |
| |
| mutex_lock(&pcm_mutex); |
| |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) |
| cpu_dai->playback.active = codec_dai->playback.active = 0; |
| else |
| cpu_dai->capture.active = codec_dai->capture.active = 0; |
| |
| if (codec_dai->playback.active == 0 && |
| codec_dai->capture.active == 0) { |
| cpu_dai->active = codec_dai->active = 0; |
| } |
| 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->ops.shutdown) |
| cpu_dai->ops.shutdown(substream, cpu_dai); |
| |
| if (codec_dai->ops.shutdown) |
| codec_dai->ops.shutdown(substream, codec_dai); |
| |
| if (machine->ops && machine->ops->shutdown) |
| machine->ops->shutdown(substream); |
| |
| if (platform->pcm_ops->close) |
| platform->pcm_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(&socdev->delayed_work, |
| msecs_to_jiffies(pmdown_time)); |
| } else { |
| /* capture streams can be powered down now */ |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->capture.stream_name, |
| SND_SOC_DAPM_STREAM_STOP); |
| |
| if (codec->active == 0 && codec_dai->pop_wait == 0) |
| snd_soc_dapm_set_bias_level(socdev, |
| SND_SOC_BIAS_STANDBY); |
| } |
| |
| 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_device *socdev = rtd->socdev; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| struct snd_soc_codec *codec = socdev->codec; |
| int ret = 0; |
| |
| mutex_lock(&pcm_mutex); |
| |
| if (machine->ops && machine->ops->prepare) { |
| ret = machine->ops->prepare(substream); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: machine prepare error\n"); |
| goto out; |
| } |
| } |
| |
| if (platform->pcm_ops->prepare) { |
| ret = platform->pcm_ops->prepare(substream); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: platform prepare error\n"); |
| goto out; |
| } |
| } |
| |
| if (codec_dai->ops.prepare) { |
| ret = codec_dai->ops.prepare(substream, codec_dai); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: codec DAI prepare error\n"); |
| goto out; |
| } |
| } |
| |
| if (cpu_dai->ops.prepare) { |
| ret = cpu_dai->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(&socdev->delayed_work); |
| } |
| |
| /* do we need to power up codec */ |
| if (codec->bias_level != SND_SOC_BIAS_ON) { |
| snd_soc_dapm_set_bias_level(socdev, |
| SND_SOC_BIAS_PREPARE); |
| |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->playback.stream_name, |
| SND_SOC_DAPM_STREAM_START); |
| else |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->capture.stream_name, |
| SND_SOC_DAPM_STREAM_START); |
| |
| snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); |
| snd_soc_dai_digital_mute(codec_dai, 0); |
| |
| } else { |
| /* codec already powered - power on widgets */ |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->playback.stream_name, |
| SND_SOC_DAPM_STREAM_START); |
| else |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->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_device *socdev = rtd->socdev; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| int ret = 0; |
| |
| mutex_lock(&pcm_mutex); |
| |
| if (machine->ops && machine->ops->hw_params) { |
| ret = machine->ops->hw_params(substream, params); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: machine hw_params failed\n"); |
| goto out; |
| } |
| } |
| |
| if (codec_dai->ops.hw_params) { |
| ret = codec_dai->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->ops.hw_params) { |
| ret = cpu_dai->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->pcm_ops->hw_params) { |
| ret = platform->pcm_ops->hw_params(substream, params); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: platform %s hw params failed\n", |
| platform->name); |
| goto platform_err; |
| } |
| } |
| |
| out: |
| mutex_unlock(&pcm_mutex); |
| return ret; |
| |
| platform_err: |
| if (cpu_dai->ops.hw_free) |
| cpu_dai->ops.hw_free(substream, cpu_dai); |
| |
| interface_err: |
| if (codec_dai->ops.hw_free) |
| codec_dai->ops.hw_free(substream, codec_dai); |
| |
| codec_err: |
| if (machine->ops && machine->ops->hw_free) |
| machine->ops->hw_free(substream); |
| |
| mutex_unlock(&pcm_mutex); |
| return ret; |
| } |
| |
| /* |
| * Free's 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_device *socdev = rtd->socdev; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| struct snd_soc_codec *codec = socdev->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 (machine->ops && machine->ops->hw_free) |
| machine->ops->hw_free(substream); |
| |
| /* free any DMA resources */ |
| if (platform->pcm_ops->hw_free) |
| platform->pcm_ops->hw_free(substream); |
| |
| /* now free hw params for the DAI's */ |
| if (codec_dai->ops.hw_free) |
| codec_dai->ops.hw_free(substream, codec_dai); |
| |
| if (cpu_dai->ops.hw_free) |
| cpu_dai->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_device *socdev = rtd->socdev; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| int ret; |
| |
| if (codec_dai->ops.trigger) { |
| ret = codec_dai->ops.trigger(substream, cmd, codec_dai); |
| if (ret < 0) |
| return ret; |
| } |
| |
| if (platform->pcm_ops->trigger) { |
| ret = platform->pcm_ops->trigger(substream, cmd); |
| if (ret < 0) |
| return ret; |
| } |
| |
| if (cpu_dai->ops.trigger) { |
| ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai); |
| if (ret < 0) |
| return ret; |
| } |
| return 0; |
| } |
| |
| /* 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, |
| }; |
| |
| #ifdef CONFIG_PM |
| /* powers down audio subsystem for suspend */ |
| static int soc_suspend(struct platform_device *pdev, pm_message_t state) |
| { |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_codec_device *codec_dev = socdev->codec_dev; |
| struct snd_soc_codec *codec = socdev->codec; |
| int i; |
| |
| /* Due to the resume being scheduled into a workqueue we could |
| * suspend before that's finished - wait for it to complete. |
| */ |
| snd_power_lock(codec->card); |
| snd_power_wait(codec->card, SNDRV_CTL_POWER_D0); |
| snd_power_unlock(codec->card); |
| |
| /* we're going to block userspace touching us until resume completes */ |
| snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); |
| |
| /* mute any active DAC's */ |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *dai = card->dai_link[i].codec_dai; |
| if (dai->ops.digital_mute && dai->playback.active) |
| dai->ops.digital_mute(dai, 1); |
| } |
| |
| /* suspend all pcms */ |
| for (i = 0; i < card->num_links; i++) |
| snd_pcm_suspend_all(card->dai_link[i].pcm); |
| |
| if (card->suspend_pre) |
| card->suspend_pre(pdev, state); |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->suspend && !cpu_dai->ac97_control) |
| cpu_dai->suspend(pdev, cpu_dai); |
| if (platform->suspend) |
| platform->suspend(pdev, cpu_dai); |
| } |
| |
| /* close any waiting streams and save state */ |
| run_delayed_work(&socdev->delayed_work); |
| codec->suspend_bias_level = codec->bias_level; |
| |
| for (i = 0; i < codec->num_dai; i++) { |
| char *stream = codec->dai[i].playback.stream_name; |
| if (stream != NULL) |
| snd_soc_dapm_stream_event(codec, stream, |
| SND_SOC_DAPM_STREAM_SUSPEND); |
| stream = codec->dai[i].capture.stream_name; |
| if (stream != NULL) |
| snd_soc_dapm_stream_event(codec, stream, |
| SND_SOC_DAPM_STREAM_SUSPEND); |
| } |
| |
| if (codec_dev->suspend) |
| codec_dev->suspend(pdev, state); |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->suspend && cpu_dai->ac97_control) |
| cpu_dai->suspend(pdev, cpu_dai); |
| } |
| |
| if (card->suspend_post) |
| card->suspend_post(pdev, state); |
| |
| return 0; |
| } |
| |
| /* deferred resume work, so resume can complete before we finished |
| * setting our codec back up, which can be very slow on I2C |
| */ |
| static void soc_resume_deferred(struct work_struct *work) |
| { |
| struct snd_soc_device *socdev = container_of(work, |
| struct snd_soc_device, |
| deferred_resume_work); |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_codec_device *codec_dev = socdev->codec_dev; |
| struct snd_soc_codec *codec = socdev->codec; |
| struct platform_device *pdev = to_platform_device(socdev->dev); |
| int i; |
| |
| /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, |
| * so userspace apps are blocked from touching us |
| */ |
| |
| dev_dbg(socdev->dev, "starting resume work\n"); |
| |
| if (card->resume_pre) |
| card->resume_pre(pdev); |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->resume && cpu_dai->ac97_control) |
| cpu_dai->resume(pdev, cpu_dai); |
| } |
| |
| if (codec_dev->resume) |
| codec_dev->resume(pdev); |
| |
| for (i = 0; i < codec->num_dai; i++) { |
| char *stream = codec->dai[i].playback.stream_name; |
| if (stream != NULL) |
| snd_soc_dapm_stream_event(codec, stream, |
| SND_SOC_DAPM_STREAM_RESUME); |
| stream = codec->dai[i].capture.stream_name; |
| if (stream != NULL) |
| snd_soc_dapm_stream_event(codec, stream, |
| SND_SOC_DAPM_STREAM_RESUME); |
| } |
| |
| /* unmute any active DACs */ |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *dai = card->dai_link[i].codec_dai; |
| if (dai->ops.digital_mute && dai->playback.active) |
| dai->ops.digital_mute(dai, 0); |
| } |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->resume && !cpu_dai->ac97_control) |
| cpu_dai->resume(pdev, cpu_dai); |
| if (platform->resume) |
| platform->resume(pdev, cpu_dai); |
| } |
| |
| if (card->resume_post) |
| card->resume_post(pdev); |
| |
| dev_dbg(socdev->dev, "resume work completed\n"); |
| |
| /* userspace can access us now we are back as we were before */ |
| snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); |
| } |
| |
| /* powers up audio subsystem after a suspend */ |
| static int soc_resume(struct platform_device *pdev) |
| { |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| |
| dev_dbg(socdev->dev, "scheduling resume work\n"); |
| |
| if (!schedule_work(&socdev->deferred_resume_work)) |
| dev_err(socdev->dev, "resume work item may be lost\n"); |
| |
| return 0; |
| } |
| |
| #else |
| #define soc_suspend NULL |
| #define soc_resume NULL |
| #endif |
| |
| /* probes a new socdev */ |
| static int soc_probe(struct platform_device *pdev) |
| { |
| int ret = 0, i; |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_codec_device *codec_dev = socdev->codec_dev; |
| |
| if (card->probe) { |
| ret = card->probe(pdev); |
| if (ret < 0) |
| return ret; |
| } |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->probe) { |
| ret = cpu_dai->probe(pdev, cpu_dai); |
| if (ret < 0) |
| goto cpu_dai_err; |
| } |
| } |
| |
| if (codec_dev->probe) { |
| ret = codec_dev->probe(pdev); |
| if (ret < 0) |
| goto cpu_dai_err; |
| } |
| |
| if (platform->probe) { |
| ret = platform->probe(pdev); |
| if (ret < 0) |
| goto platform_err; |
| } |
| |
| /* DAPM stream work */ |
| INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); |
| #ifdef CONFIG_PM |
| /* deferred resume work */ |
| INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); |
| #endif |
| |
| return 0; |
| |
| platform_err: |
| if (codec_dev->remove) |
| codec_dev->remove(pdev); |
| |
| cpu_dai_err: |
| for (i--; i >= 0; i--) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->remove) |
| cpu_dai->remove(pdev, cpu_dai); |
| } |
| |
| if (card->remove) |
| card->remove(pdev); |
| |
| return ret; |
| } |
| |
| /* removes a socdev */ |
| static int soc_remove(struct platform_device *pdev) |
| { |
| int i; |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_platform *platform = socdev->platform; |
| struct snd_soc_codec_device *codec_dev = socdev->codec_dev; |
| |
| run_delayed_work(&socdev->delayed_work); |
| |
| if (platform->remove) |
| platform->remove(pdev); |
| |
| if (codec_dev->remove) |
| codec_dev->remove(pdev); |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->remove) |
| cpu_dai->remove(pdev, cpu_dai); |
| } |
| |
| if (card->remove) |
| card->remove(pdev); |
| |
| return 0; |
| } |
| |
| /* ASoC platform driver */ |
| static struct platform_driver soc_driver = { |
| .driver = { |
| .name = "soc-audio", |
| .owner = THIS_MODULE, |
| }, |
| .probe = soc_probe, |
| .remove = soc_remove, |
| .suspend = soc_suspend, |
| .resume = soc_resume, |
| }; |
| |
| /* create a new pcm */ |
| static int soc_new_pcm(struct snd_soc_device *socdev, |
| struct snd_soc_dai_link *dai_link, int num) |
| { |
| struct snd_soc_codec *codec = socdev->codec; |
| struct snd_soc_dai *codec_dai = dai_link->codec_dai; |
| struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; |
| struct snd_soc_pcm_runtime *rtd; |
| struct snd_pcm *pcm; |
| char new_name[64]; |
| int ret = 0, playback = 0, capture = 0; |
| |
| rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); |
| if (rtd == NULL) |
| return -ENOMEM; |
| |
| rtd->dai = dai_link; |
| rtd->socdev = socdev; |
| codec_dai->codec = socdev->codec; |
| |
| /* check client and interface hw capabilities */ |
| sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name, |
| num); |
| |
| if (codec_dai->playback.channels_min) |
| playback = 1; |
| if (codec_dai->capture.channels_min) |
| capture = 1; |
| |
| ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, |
| capture, &pcm); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't create pcm for codec %s\n", |
| codec->name); |
| kfree(rtd); |
| return ret; |
| } |
| |
| dai_link->pcm = pcm; |
| pcm->private_data = rtd; |
| soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap; |
| soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer; |
| soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl; |
| soc_pcm_ops.copy = socdev->platform->pcm_ops->copy; |
| soc_pcm_ops.silence = socdev->platform->pcm_ops->silence; |
| soc_pcm_ops.ack = socdev->platform->pcm_ops->ack; |
| soc_pcm_ops.page = socdev->platform->pcm_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); |
| |
| ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: platform pcm constructor failed\n"); |
| kfree(rtd); |
| return ret; |
| } |
| |
| pcm->private_free = socdev->platform->pcm_free; |
| printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, |
| cpu_dai->name); |
| return ret; |
| } |
| |
| /* codec register dump */ |
| static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) |
| { |
| struct snd_soc_codec *codec = devdata->codec; |
| int i, step = 1, count = 0; |
| |
| if (!codec->reg_cache_size) |
| return 0; |
| |
| if (codec->reg_cache_step) |
| step = codec->reg_cache_step; |
| |
| count += sprintf(buf, "%s registers\n", codec->name); |
| for (i = 0; i < codec->reg_cache_size; i += step) { |
| count += sprintf(buf + count, "%2x: ", i); |
| if (count >= PAGE_SIZE - 1) |
| break; |
| |
| if (codec->display_register) |
| count += codec->display_register(codec, buf + count, |
| PAGE_SIZE - count, i); |
| else |
| count += snprintf(buf + count, PAGE_SIZE - count, |
| "%4x", codec->read(codec, i)); |
| |
| if (count >= PAGE_SIZE - 1) |
| break; |
| |
| count += snprintf(buf + count, PAGE_SIZE - count, "\n"); |
| if (count >= PAGE_SIZE - 1) |
| break; |
| } |
| |
| /* Truncate count; min() would cause a warning */ |
| if (count >= PAGE_SIZE) |
| count = PAGE_SIZE - 1; |
| |
| return count; |
| } |
| static ssize_t codec_reg_show(struct device *dev, |
| struct device_attribute *attr, char *buf) |
| { |
| struct snd_soc_device *devdata = dev_get_drvdata(dev); |
| return soc_codec_reg_show(devdata, buf); |
| } |
| |
| static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); |
| |
| #ifdef CONFIG_DEBUG_FS |
| static int codec_reg_open_file(struct inode *inode, struct file *file) |
| { |
| file->private_data = inode->i_private; |
| return 0; |
| } |
| |
| static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, |
| size_t count, loff_t *ppos) |
| { |
| ssize_t ret; |
| struct snd_soc_device *devdata = file->private_data; |
| char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); |
| if (!buf) |
| return -ENOMEM; |
| ret = soc_codec_reg_show(devdata, buf); |
| if (ret >= 0) |
| ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); |
| kfree(buf); |
| return ret; |
| } |
| |
| static ssize_t codec_reg_write_file(struct file *file, |
| const char __user *user_buf, size_t count, loff_t *ppos) |
| { |
| char buf[32]; |
| int buf_size; |
| char *start = buf; |
| unsigned long reg, value; |
| int step = 1; |
| struct snd_soc_device *devdata = file->private_data; |
| struct snd_soc_codec *codec = devdata->codec; |
| |
| buf_size = min(count, (sizeof(buf)-1)); |
| if (copy_from_user(buf, user_buf, buf_size)) |
| return -EFAULT; |
| buf[buf_size] = 0; |
| |
| if (codec->reg_cache_step) |
| step = codec->reg_cache_step; |
| |
| while (*start == ' ') |
| start++; |
| reg = simple_strtoul(start, &start, 16); |
| if ((reg >= codec->reg_cache_size) || (reg % step)) |
| return -EINVAL; |
| while (*start == ' ') |
| start++; |
| if (strict_strtoul(start, 16, &value)) |
| return -EINVAL; |
| codec->write(codec, reg, value); |
| return buf_size; |
| } |
| |
| static const struct file_operations codec_reg_fops = { |
| .open = codec_reg_open_file, |
| .read = codec_reg_read_file, |
| .write = codec_reg_write_file, |
| }; |
| |
| static void soc_init_debugfs(struct snd_soc_device *socdev) |
| { |
| struct dentry *root, *file; |
| struct snd_soc_codec *codec = socdev->codec; |
| root = debugfs_create_dir(dev_name(socdev->dev), NULL); |
| if (IS_ERR(root) || !root) |
| goto exit1; |
| |
| file = debugfs_create_file("codec_reg", 0644, |
| root, socdev, &codec_reg_fops); |
| if (!file) |
| goto exit2; |
| |
| file = debugfs_create_u32("dapm_pop_time", 0744, |
| root, &codec->pop_time); |
| if (!file) |
| goto exit2; |
| socdev->debugfs_root = root; |
| return; |
| exit2: |
| debugfs_remove_recursive(root); |
| exit1: |
| dev_err(socdev->dev, "debugfs is not available\n"); |
| } |
| |
| static void soc_cleanup_debugfs(struct snd_soc_device *socdev) |
| { |
| debugfs_remove_recursive(socdev->debugfs_root); |
| socdev->debugfs_root = NULL; |
| } |
| |
| #else |
| |
| static inline void soc_init_debugfs(struct snd_soc_device *socdev) |
| { |
| } |
| |
| static inline void soc_cleanup_debugfs(struct snd_soc_device *socdev) |
| { |
| } |
| #endif |
| |
| /** |
| * snd_soc_new_ac97_codec - initailise AC97 device |
| * @codec: audio codec |
| * @ops: AC97 bus operations |
| * @num: AC97 codec number |
| * |
| * Initialises AC97 codec resources for use by ad-hoc devices only. |
| */ |
| int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, |
| struct snd_ac97_bus_ops *ops, int num) |
| { |
| mutex_lock(&codec->mutex); |
| |
| codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); |
| if (codec->ac97 == NULL) { |
| mutex_unlock(&codec->mutex); |
| return -ENOMEM; |
| } |
| |
| codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); |
| if (codec->ac97->bus == NULL) { |
| kfree(codec->ac97); |
| codec->ac97 = NULL; |
| mutex_unlock(&codec->mutex); |
| return -ENOMEM; |
| } |
| |
| codec->ac97->bus->ops = ops; |
| codec->ac97->num = num; |
| mutex_unlock(&codec->mutex); |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); |
| |
| /** |
| * snd_soc_free_ac97_codec - free AC97 codec device |
| * @codec: audio codec |
| * |
| * Frees AC97 codec device resources. |
| */ |
| void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) |
| { |
| mutex_lock(&codec->mutex); |
| kfree(codec->ac97->bus); |
| kfree(codec->ac97); |
| codec->ac97 = NULL; |
| mutex_unlock(&codec->mutex); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); |
| |
| /** |
| * snd_soc_update_bits - update codec register bits |
| * @codec: audio codec |
| * @reg: codec register |
| * @mask: register mask |
| * @value: new value |
| * |
| * Writes new register value. |
| * |
| * Returns 1 for change else 0. |
| */ |
| int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, |
| unsigned short mask, unsigned short value) |
| { |
| int change; |
| unsigned short old, new; |
| |
| mutex_lock(&io_mutex); |
| old = snd_soc_read(codec, reg); |
| new = (old & ~mask) | value; |
| change = old != new; |
| if (change) |
| snd_soc_write(codec, reg, new); |
| |
| mutex_unlock(&io_mutex); |
| return change; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_update_bits); |
| |
| /** |
| * snd_soc_test_bits - test register for change |
| * @codec: audio codec |
| * @reg: codec register |
| * @mask: register mask |
| * @value: new value |
| * |
| * Tests a register with a new value and checks if the new value is |
| * different from the old value. |
| * |
| * Returns 1 for change else 0. |
| */ |
| int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, |
| unsigned short mask, unsigned short value) |
| { |
| int change; |
| unsigned short old, new; |
| |
| mutex_lock(&io_mutex); |
| old = snd_soc_read(codec, reg); |
| new = (old & ~mask) | value; |
| change = old != new; |
| mutex_unlock(&io_mutex); |
| |
| return change; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_test_bits); |
| |
| /** |
| * snd_soc_new_pcms - create new sound card and pcms |
| * @socdev: the SoC audio device |
| * |
| * Create a new sound card based upon the codec and interface pcms. |
| * |
| * Returns 0 for success, else error. |
| */ |
| int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) |
| { |
| struct snd_soc_codec *codec = socdev->codec; |
| struct snd_soc_card *card = socdev->card; |
| int ret = 0, i; |
| |
| mutex_lock(&codec->mutex); |
| |
| /* register a sound card */ |
| codec->card = snd_card_new(idx, xid, codec->owner, 0); |
| if (!codec->card) { |
| printk(KERN_ERR "asoc: can't create sound card for codec %s\n", |
| codec->name); |
| mutex_unlock(&codec->mutex); |
| return -ENODEV; |
| } |
| |
| codec->card->dev = socdev->dev; |
| codec->card->private_data = codec; |
| strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); |
| |
| /* create the pcms */ |
| for (i = 0; i < card->num_links; i++) { |
| ret = soc_new_pcm(socdev, &card->dai_link[i], i); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't create pcm %s\n", |
| card->dai_link[i].stream_name); |
| mutex_unlock(&codec->mutex); |
| return ret; |
| } |
| } |
| |
| mutex_unlock(&codec->mutex); |
| return ret; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_new_pcms); |
| |
| /** |
| * snd_soc_init_card - register sound card |
| * @socdev: the SoC audio device |
| * |
| * Register a SoC sound card. Also registers an AC97 device if the |
| * codec is AC97 for ad hoc devices. |
| * |
| * Returns 0 for success, else error. |
| */ |
| int snd_soc_init_card(struct snd_soc_device *socdev) |
| { |
| struct snd_soc_codec *codec = socdev->codec; |
| struct snd_soc_card *card = socdev->card; |
| int ret = 0, i, ac97 = 0, err = 0; |
| |
| for (i = 0; i < card->num_links; i++) { |
| if (card->dai_link[i].init) { |
| err = card->dai_link[i].init(codec); |
| if (err < 0) { |
| printk(KERN_ERR "asoc: failed to init %s\n", |
| card->dai_link[i].stream_name); |
| continue; |
| } |
| } |
| if (card->dai_link[i].codec_dai->ac97_control) |
| ac97 = 1; |
| } |
| snprintf(codec->card->shortname, sizeof(codec->card->shortname), |
| "%s", card->name); |
| snprintf(codec->card->longname, sizeof(codec->card->longname), |
| "%s (%s)", card->name, codec->name); |
| |
| ret = snd_card_register(codec->card); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: failed to register soundcard for %s\n", |
| codec->name); |
| goto out; |
| } |
| |
| mutex_lock(&codec->mutex); |
| #ifdef CONFIG_SND_SOC_AC97_BUS |
| if (ac97) { |
| ret = soc_ac97_dev_register(codec); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: AC97 device register failed\n"); |
| snd_card_free(codec->card); |
| mutex_unlock(&codec->mutex); |
| goto out; |
| } |
| } |
| #endif |
| |
| err = snd_soc_dapm_sys_add(socdev->dev); |
| if (err < 0) |
| printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); |
| |
| err = device_create_file(socdev->dev, &dev_attr_codec_reg); |
| if (err < 0) |
| printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); |
| |
| soc_init_debugfs(socdev); |
| mutex_unlock(&codec->mutex); |
| |
| out: |
| return ret; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_init_card); |
| |
| /** |
| * snd_soc_free_pcms - free sound card and pcms |
| * @socdev: the SoC audio device |
| * |
| * Frees sound card and pcms associated with the socdev. |
| * Also unregister the codec if it is an AC97 device. |
| */ |
| void snd_soc_free_pcms(struct snd_soc_device *socdev) |
| { |
| struct snd_soc_codec *codec = socdev->codec; |
| #ifdef CONFIG_SND_SOC_AC97_BUS |
| struct snd_soc_dai *codec_dai; |
| int i; |
| #endif |
| |
| mutex_lock(&codec->mutex); |
| soc_cleanup_debugfs(socdev); |
| #ifdef CONFIG_SND_SOC_AC97_BUS |
| for (i = 0; i < codec->num_dai; i++) { |
| codec_dai = &codec->dai[i]; |
| if (codec_dai->ac97_control && codec->ac97) { |
| soc_ac97_dev_unregister(codec); |
| goto free_card; |
| } |
| } |
| free_card: |
| #endif |
| |
| if (codec->card) |
| snd_card_free(codec->card); |
| device_remove_file(socdev->dev, &dev_attr_codec_reg); |
| mutex_unlock(&codec->mutex); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_free_pcms); |
| |
| /** |
| * snd_soc_set_runtime_hwparams - set the runtime hardware parameters |
| * @substream: the pcm substream |
| * @hw: the hardware parameters |
| * |
| * Sets the substream runtime hardware parameters. |
| */ |
| int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, |
| const struct snd_pcm_hardware *hw) |
| { |
| struct snd_pcm_runtime *runtime = substream->runtime; |
| runtime->hw.info = hw->info; |
| runtime->hw.formats = hw->formats; |
| runtime->hw.period_bytes_min = hw->period_bytes_min; |
| runtime->hw.period_bytes_max = hw->period_bytes_max; |
| runtime->hw.periods_min = hw->periods_min; |
| runtime->hw.periods_max = hw->periods_max; |
| runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; |
| runtime->hw.fifo_size = hw->fifo_size; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); |
| |
| /** |
| * snd_soc_cnew - create new control |
| * @_template: control template |
| * @data: control private data |
| * @lnng_name: control long name |
| * |
| * Create a new mixer control from a template control. |
| * |
| * Returns 0 for success, else error. |
| */ |
| struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, |
| void *data, char *long_name) |
| { |
| struct snd_kcontrol_new template; |
| |
| memcpy(&template, _template, sizeof(template)); |
| if (long_name) |
| template.name = long_name; |
| template.index = 0; |
| |
| return snd_ctl_new1(&template, data); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_cnew); |
| |
| /** |
| * snd_soc_info_enum_double - enumerated double mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a double enumerated |
| * mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; |
| |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; |
| uinfo->count = e->shift_l == e->shift_r ? 1 : 2; |
| uinfo->value.enumerated.items = e->max; |
| |
| if (uinfo->value.enumerated.item > e->max - 1) |
| uinfo->value.enumerated.item = e->max - 1; |
| strcpy(uinfo->value.enumerated.name, |
| e->texts[uinfo->value.enumerated.item]); |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); |
| |
| /** |
| * snd_soc_get_enum_double - enumerated double mixer get callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to get the value of a double enumerated mixer. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; |
| unsigned short val, bitmask; |
| |
| for (bitmask = 1; bitmask < e->max; bitmask <<= 1) |
| ; |
| val = snd_soc_read(codec, e->reg); |
| ucontrol->value.enumerated.item[0] |
| = (val >> e->shift_l) & (bitmask - 1); |
| if (e->shift_l != e->shift_r) |
| ucontrol->value.enumerated.item[1] = |
| (val >> e->shift_r) & (bitmask - 1); |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); |
| |
| /** |
| * snd_soc_put_enum_double - enumerated double mixer put callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to set the value of a double enumerated mixer. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; |
| unsigned short val; |
| unsigned short mask, bitmask; |
| |
| for (bitmask = 1; bitmask < e->max; bitmask <<= 1) |
| ; |
| if (ucontrol->value.enumerated.item[0] > e->max - 1) |
| return -EINVAL; |
| val = ucontrol->value.enumerated.item[0] << e->shift_l; |
| mask = (bitmask - 1) << e->shift_l; |
| if (e->shift_l != e->shift_r) { |
| if (ucontrol->value.enumerated.item[1] > e->max - 1) |
| return -EINVAL; |
| val |= ucontrol->value.enumerated.item[1] << e->shift_r; |
| mask |= (bitmask - 1) << e->shift_r; |
| } |
| |
| return snd_soc_update_bits(codec, e->reg, mask, val); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); |
| |
| /** |
| * snd_soc_info_enum_ext - external enumerated single mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about an external enumerated |
| * single mixer. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; |
| |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; |
| uinfo->count = 1; |
| uinfo->value.enumerated.items = e->max; |
| |
| if (uinfo->value.enumerated.item > e->max - 1) |
| uinfo->value.enumerated.item = e->max - 1; |
| strcpy(uinfo->value.enumerated.name, |
| e->texts[uinfo->value.enumerated.item]); |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); |
| |
| /** |
| * snd_soc_info_volsw_ext - external single mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a single external mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| int max = kcontrol->private_value; |
| |
| if (max == 1) |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; |
| else |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| |
| uinfo->count = 1; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = max; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); |
| |
| /** |
| * snd_soc_info_volsw - single mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a single mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| int max = mc->max; |
| unsigned int shift = mc->shift; |
| unsigned int rshift = mc->rshift; |
| |
| if (max == 1) |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; |
| else |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| |
| uinfo->count = shift == rshift ? 1 : 2; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = max; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_volsw); |
| |
| /** |
| * snd_soc_get_volsw - single mixer get callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to get the value of a single mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| unsigned int shift = mc->shift; |
| unsigned int rshift = mc->rshift; |
| int max = mc->max; |
| unsigned int mask = (1 << fls(max)) - 1; |
| unsigned int invert = mc->invert; |
| |
| ucontrol->value.integer.value[0] = |
| (snd_soc_read(codec, reg) >> shift) & mask; |
| if (shift != rshift) |
| ucontrol->value.integer.value[1] = |
| (snd_soc_read(codec, reg) >> rshift) & mask; |
| if (invert) { |
| ucontrol->value.integer.value[0] = |
| max - ucontrol->value.integer.value[0]; |
| if (shift != rshift) |
| ucontrol->value.integer.value[1] = |
| max - ucontrol->value.integer.value[1]; |
| } |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_volsw); |
| |
| /** |
| * snd_soc_put_volsw - single mixer put callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to set the value of a single mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| unsigned int shift = mc->shift; |
| unsigned int rshift = mc->rshift; |
| int max = mc->max; |
| unsigned int mask = (1 << fls(max)) - 1; |
| unsigned int invert = mc->invert; |
| unsigned short val, val2, val_mask; |
| |
| val = (ucontrol->value.integer.value[0] & mask); |
| if (invert) |
| val = max - val; |
| val_mask = mask << shift; |
| val = val << shift; |
| if (shift != rshift) { |
| val2 = (ucontrol->value.integer.value[1] & mask); |
| if (invert) |
| val2 = max - val2; |
| val_mask |= mask << rshift; |
| val |= val2 << rshift; |
| } |
| return snd_soc_update_bits(codec, reg, val_mask, val); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_volsw); |
| |
| /** |
| * snd_soc_info_volsw_2r - double mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a double mixer control that |
| * spans 2 codec registers. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| int max = mc->max; |
| |
| if (max == 1) |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; |
| else |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| |
| uinfo->count = 2; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = max; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); |
| |
| /** |
| * snd_soc_get_volsw_2r - double mixer get callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to get the value of a double mixer control that spans 2 registers. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| unsigned int reg2 = mc->rreg; |
| unsigned int shift = mc->shift; |
| int max = mc->max; |
| unsigned int mask = (1<<fls(max))-1; |
| unsigned int invert = mc->invert; |
| |
| ucontrol->value.integer.value[0] = |
| (snd_soc_read(codec, reg) >> shift) & mask; |
| ucontrol->value.integer.value[1] = |
| (snd_soc_read(codec, reg2) >> shift) & mask; |
| if (invert) { |
| ucontrol->value.integer.value[0] = |
| max - ucontrol->value.integer.value[0]; |
| ucontrol->value.integer.value[1] = |
| max - ucontrol->value.integer.value[1]; |
| } |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); |
| |
| /** |
| * snd_soc_put_volsw_2r - double mixer set callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to set the value of a double mixer control that spans 2 registers. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| unsigned int reg2 = mc->rreg; |
| unsigned int shift = mc->shift; |
| int max = mc->max; |
| unsigned int mask = (1 << fls(max)) - 1; |
| unsigned int invert = mc->invert; |
| int err; |
| unsigned short val, val2, val_mask; |
| |
| val_mask = mask << shift; |
| val = (ucontrol->value.integer.value[0] & mask); |
| val2 = (ucontrol->value.integer.value[1] & mask); |
| |
| if (invert) { |
| val = max - val; |
| val2 = max - val2; |
| } |
| |
| val = val << shift; |
| val2 = val2 << shift; |
| |
| err = snd_soc_update_bits(codec, reg, val_mask, val); |
| if (err < 0) |
| return err; |
| |
| err = snd_soc_update_bits(codec, reg2, val_mask, val2); |
| return err; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); |
| |
| /** |
| * snd_soc_info_volsw_s8 - signed mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a signed mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| int max = mc->max; |
| int min = mc->min; |
| |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| uinfo->count = 2; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = max-min; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); |
| |
| /** |
| * snd_soc_get_volsw_s8 - signed mixer get callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to get the value of a signed mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| int min = mc->min; |
| int val = snd_soc_read(codec, reg); |
| |
| ucontrol->value.integer.value[0] = |
| ((signed char)(val & 0xff))-min; |
| ucontrol->value.integer.value[1] = |
| ((signed char)((val >> 8) & 0xff))-min; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); |
| |
| /** |
| * snd_soc_put_volsw_sgn - signed mixer put callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to set the value of a signed mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| int min = mc->min; |
| unsigned short val; |
| |
| val = (ucontrol->value.integer.value[0]+min) & 0xff; |
| val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; |
| |
| return snd_soc_update_bits(codec, reg, 0xffff, val); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); |
| |
| /** |
| * snd_soc_dai_set_sysclk - configure DAI system or master clock. |
| * @dai: DAI |
| * @clk_id: DAI specific clock ID |
| * @freq: new clock frequency in Hz |
| * @dir: new clock direction - input/output. |
| * |
| * Configures the DAI master (MCLK) or system (SYSCLK) clocking. |
| */ |
| int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, |
| unsigned int freq, int dir) |
| { |
| if (dai->ops.set_sysclk) |
| return dai->ops.set_sysclk(dai, clk_id, freq, dir); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); |
| |
| /** |
| * snd_soc_dai_set_clkdiv - configure DAI clock dividers. |
| * @dai: DAI |
| * @clk_id: DAI specific clock divider ID |
| * @div: new clock divisor. |
| * |
| * Configures the clock dividers. This is used to derive the best DAI bit and |
| * frame clocks from the system or master clock. It's best to set the DAI bit |
| * and frame clocks as low as possible to save system power. |
| */ |
| int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, |
| int div_id, int div) |
| { |
| if (dai->ops.set_clkdiv) |
| return dai->ops.set_clkdiv(dai, div_id, div); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); |
| |
| /** |
| * snd_soc_dai_set_pll - configure DAI PLL. |
| * @dai: DAI |
| * @pll_id: DAI specific PLL ID |
| * @freq_in: PLL input clock frequency in Hz |
| * @freq_out: requested PLL output clock frequency in Hz |
| * |
| * Configures and enables PLL to generate output clock based on input clock. |
| */ |
| int snd_soc_dai_set_pll(struct snd_soc_dai *dai, |
| int pll_id, unsigned int freq_in, unsigned int freq_out) |
| { |
| if (dai->ops.set_pll) |
| return dai->ops.set_pll(dai, pll_id, freq_in, freq_out); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); |
| |
| /** |
| * snd_soc_dai_set_fmt - configure DAI hardware audio format. |
| * @dai: DAI |
| * @fmt: SND_SOC_DAIFMT_ format value. |
| * |
| * Configures the DAI hardware format and clocking. |
| */ |
| int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) |
| { |
| if (dai->ops.set_fmt) |
| return dai->ops.set_fmt(dai, fmt); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); |
| |
| /** |
| * snd_soc_dai_set_tdm_slot - configure DAI TDM. |
| * @dai: DAI |
| * @mask: DAI specific mask representing used slots. |
| * @slots: Number of slots in use. |
| * |
| * Configures a DAI for TDM operation. Both mask and slots are codec and DAI |
| * specific. |
| */ |
| int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, |
| unsigned int mask, int slots) |
| { |
| if (dai->ops.set_sysclk) |
| return dai->ops.set_tdm_slot(dai, mask, slots); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); |
| |
| /** |
| * snd_soc_dai_set_tristate - configure DAI system or master clock. |
| * @dai: DAI |
| * @tristate: tristate enable |
| * |
| * Tristates the DAI so that others can use it. |
| */ |
| int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) |
| { |
| if (dai->ops.set_sysclk) |
| return dai->ops.set_tristate(dai, tristate); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); |
| |
| /** |
| * snd_soc_dai_digital_mute - configure DAI system or master clock. |
| * @dai: DAI |
| * @mute: mute enable |
| * |
| * Mutes the DAI DAC. |
| */ |
| int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) |
| { |
| if (dai->ops.digital_mute) |
| return dai->ops.digital_mute(dai, mute); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); |
| |
| static int __devinit snd_soc_init(void) |
| { |
| return platform_driver_register(&soc_driver); |
| } |
| |
| static void snd_soc_exit(void) |
| { |
| platform_driver_unregister(&soc_driver); |
| } |
| |
| module_init(snd_soc_init); |
| module_exit(snd_soc_exit); |
| |
| /* Module information */ |
| MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); |
| MODULE_DESCRIPTION("ALSA SoC Core"); |
| MODULE_LICENSE("GPL"); |
| MODULE_ALIAS("platform:soc-audio"); |