/* * soc-core.c -- ALSA SoC Audio Layer * * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * Author: Liam Girdwood * with code, comments and ideas from :- * Richard Purdie * * 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 #include #include #include #include #include #include #include #include #include #include #include #include #include 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<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 __exit 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");