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875065491f
One of the issues with the ASoC v1 API which has been addressed in the ASoC v2 work that Liam Girdwood has done is that the ALSA card provided by ASoC is distributed around the ASoC structures. For example, machine wide data such as the struct snd_card are maintained as part of the CODEC data structure, preventing the use of multiple codecs. This has been addressed by refactoring the data structures so that all the data for the ALSA card is contained in a single structure snd_soc_card which replaces the existing snd_soc_machine and snd_soc_device. Begin the process of backporting this by renaming struct snd_soc_machine to struct snd_soc_card, better reflecting its function and bringing it closer to standard ALSA terminology. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1984 lines
52 KiB
C
1984 lines
52 KiB
C
/*
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* soc-core.c -- ALSA SoC Audio Layer
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*
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* Copyright 2005 Wolfson Microelectronics PLC.
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* Copyright 2005 Openedhand Ltd.
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*
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* Author: Liam Girdwood <lrg@slimlogic.co.uk>
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* with code, comments and ideas from :-
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* Richard Purdie <richard@openedhand.com>
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU General Public License as published by the
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* Free Software Foundation; either version 2 of the License, or (at your
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* option) any later version.
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*
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* TODO:
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* o Add hw rules to enforce rates, etc.
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* o More testing with other codecs/machines.
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* o Add more codecs and platforms to ensure good API coverage.
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* o Support TDM on PCM and I2S
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*/
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#include <linux/module.h>
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#include <linux/moduleparam.h>
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#include <linux/init.h>
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#include <linux/delay.h>
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#include <linux/pm.h>
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#include <linux/bitops.h>
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#include <linux/debugfs.h>
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#include <linux/platform_device.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include <sound/soc-dapm.h>
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#include <sound/initval.h>
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static DEFINE_MUTEX(pcm_mutex);
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static DEFINE_MUTEX(io_mutex);
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static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
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/*
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* This is a timeout to do a DAPM powerdown after a stream is closed().
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* It can be used to eliminate pops between different playback streams, e.g.
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* between two audio tracks.
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*/
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static int pmdown_time = 5000;
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module_param(pmdown_time, int, 0);
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MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
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/*
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* This function forces any delayed work to be queued and run.
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*/
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static int run_delayed_work(struct delayed_work *dwork)
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{
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int ret;
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/* cancel any work waiting to be queued. */
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ret = cancel_delayed_work(dwork);
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/* if there was any work waiting then we run it now and
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* wait for it's completion */
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if (ret) {
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schedule_delayed_work(dwork, 0);
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flush_scheduled_work();
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}
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return ret;
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}
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#ifdef CONFIG_SND_SOC_AC97_BUS
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/* unregister ac97 codec */
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static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
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{
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if (codec->ac97->dev.bus)
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device_unregister(&codec->ac97->dev);
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return 0;
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}
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/* stop no dev release warning */
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static void soc_ac97_device_release(struct device *dev){}
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/* register ac97 codec to bus */
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static int soc_ac97_dev_register(struct snd_soc_codec *codec)
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{
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int err;
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codec->ac97->dev.bus = &ac97_bus_type;
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codec->ac97->dev.parent = NULL;
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codec->ac97->dev.release = soc_ac97_device_release;
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snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
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codec->card->number, 0, codec->name);
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err = device_register(&codec->ac97->dev);
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if (err < 0) {
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snd_printk(KERN_ERR "Can't register ac97 bus\n");
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codec->ac97->dev.bus = NULL;
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return err;
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}
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return 0;
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}
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#endif
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static inline const char *get_dai_name(int type)
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{
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switch (type) {
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case SND_SOC_DAI_AC97_BUS:
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case SND_SOC_DAI_AC97:
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return "AC97";
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case SND_SOC_DAI_I2S:
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return "I2S";
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case SND_SOC_DAI_PCM:
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return "PCM";
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}
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return NULL;
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}
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/*
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* Called by ALSA when a PCM substream is opened, the runtime->hw record is
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* then initialized and any private data can be allocated. This also calls
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* startup for the cpu DAI, platform, machine and codec DAI.
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*/
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static int soc_pcm_open(struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_device *socdev = rtd->socdev;
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct snd_soc_dai_link *machine = rtd->dai;
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struct snd_soc_platform *platform = socdev->platform;
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struct snd_soc_dai *cpu_dai = machine->cpu_dai;
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struct snd_soc_dai *codec_dai = machine->codec_dai;
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int ret = 0;
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mutex_lock(&pcm_mutex);
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/* startup the audio subsystem */
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if (cpu_dai->ops.startup) {
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ret = cpu_dai->ops.startup(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: can't open interface %s\n",
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cpu_dai->name);
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goto out;
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}
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}
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if (platform->pcm_ops->open) {
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ret = platform->pcm_ops->open(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
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goto platform_err;
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}
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}
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if (codec_dai->ops.startup) {
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ret = codec_dai->ops.startup(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: can't open codec %s\n",
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codec_dai->name);
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goto codec_dai_err;
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}
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}
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if (machine->ops && machine->ops->startup) {
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ret = machine->ops->startup(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
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goto machine_err;
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}
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}
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/* Check that the codec and cpu DAI's are compatible */
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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runtime->hw.rate_min =
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max(codec_dai->playback.rate_min,
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cpu_dai->playback.rate_min);
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runtime->hw.rate_max =
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min(codec_dai->playback.rate_max,
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cpu_dai->playback.rate_max);
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runtime->hw.channels_min =
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max(codec_dai->playback.channels_min,
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cpu_dai->playback.channels_min);
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runtime->hw.channels_max =
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min(codec_dai->playback.channels_max,
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cpu_dai->playback.channels_max);
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runtime->hw.formats =
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codec_dai->playback.formats & cpu_dai->playback.formats;
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runtime->hw.rates =
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codec_dai->playback.rates & cpu_dai->playback.rates;
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} else {
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runtime->hw.rate_min =
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max(codec_dai->capture.rate_min,
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cpu_dai->capture.rate_min);
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runtime->hw.rate_max =
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min(codec_dai->capture.rate_max,
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cpu_dai->capture.rate_max);
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runtime->hw.channels_min =
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max(codec_dai->capture.channels_min,
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cpu_dai->capture.channels_min);
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runtime->hw.channels_max =
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min(codec_dai->capture.channels_max,
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cpu_dai->capture.channels_max);
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runtime->hw.formats =
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codec_dai->capture.formats & cpu_dai->capture.formats;
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runtime->hw.rates =
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codec_dai->capture.rates & cpu_dai->capture.rates;
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}
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snd_pcm_limit_hw_rates(runtime);
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if (!runtime->hw.rates) {
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printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
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codec_dai->name, cpu_dai->name);
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goto machine_err;
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}
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if (!runtime->hw.formats) {
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printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
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codec_dai->name, cpu_dai->name);
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goto machine_err;
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}
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if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
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printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
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codec_dai->name, cpu_dai->name);
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goto machine_err;
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}
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pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
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pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
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pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
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runtime->hw.channels_max);
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pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
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runtime->hw.rate_max);
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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cpu_dai->playback.active = codec_dai->playback.active = 1;
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else
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cpu_dai->capture.active = codec_dai->capture.active = 1;
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cpu_dai->active = codec_dai->active = 1;
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cpu_dai->runtime = runtime;
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socdev->codec->active++;
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mutex_unlock(&pcm_mutex);
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return 0;
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machine_err:
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if (machine->ops && machine->ops->shutdown)
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machine->ops->shutdown(substream);
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codec_dai_err:
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if (platform->pcm_ops->close)
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platform->pcm_ops->close(substream);
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platform_err:
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if (cpu_dai->ops.shutdown)
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cpu_dai->ops.shutdown(substream);
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out:
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mutex_unlock(&pcm_mutex);
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return ret;
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}
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/*
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* Power down the audio subsystem pmdown_time msecs after close is called.
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* This is to ensure there are no pops or clicks in between any music tracks
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* due to DAPM power cycling.
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*/
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static void close_delayed_work(struct work_struct *work)
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{
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struct snd_soc_device *socdev =
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container_of(work, struct snd_soc_device, delayed_work.work);
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struct snd_soc_codec *codec = socdev->codec;
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struct snd_soc_dai *codec_dai;
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int i;
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mutex_lock(&pcm_mutex);
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for (i = 0; i < codec->num_dai; i++) {
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codec_dai = &codec->dai[i];
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pr_debug("pop wq checking: %s status: %s waiting: %s\n",
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codec_dai->playback.stream_name,
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codec_dai->playback.active ? "active" : "inactive",
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codec_dai->pop_wait ? "yes" : "no");
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/* are we waiting on this codec DAI stream */
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if (codec_dai->pop_wait == 1) {
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/* Reduce power if no longer active */
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if (codec->active == 0) {
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pr_debug("pop wq D1 %s %s\n", codec->name,
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codec_dai->playback.stream_name);
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snd_soc_dapm_set_bias_level(socdev,
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SND_SOC_BIAS_PREPARE);
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}
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codec_dai->pop_wait = 0;
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snd_soc_dapm_stream_event(codec,
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codec_dai->playback.stream_name,
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SND_SOC_DAPM_STREAM_STOP);
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/* Fall into standby if no longer active */
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if (codec->active == 0) {
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pr_debug("pop wq D3 %s %s\n", codec->name,
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codec_dai->playback.stream_name);
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snd_soc_dapm_set_bias_level(socdev,
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SND_SOC_BIAS_STANDBY);
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}
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}
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}
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mutex_unlock(&pcm_mutex);
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}
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/*
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* Called by ALSA when a PCM substream is closed. Private data can be
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* freed here. The cpu DAI, codec DAI, machine and platform are also
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* shutdown.
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*/
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static int soc_codec_close(struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_device *socdev = rtd->socdev;
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struct snd_soc_dai_link *machine = rtd->dai;
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struct snd_soc_platform *platform = socdev->platform;
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struct snd_soc_dai *cpu_dai = machine->cpu_dai;
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struct snd_soc_dai *codec_dai = machine->codec_dai;
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struct snd_soc_codec *codec = socdev->codec;
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mutex_lock(&pcm_mutex);
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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cpu_dai->playback.active = codec_dai->playback.active = 0;
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else
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cpu_dai->capture.active = codec_dai->capture.active = 0;
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if (codec_dai->playback.active == 0 &&
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codec_dai->capture.active == 0) {
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cpu_dai->active = codec_dai->active = 0;
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}
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codec->active--;
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/* Muting the DAC suppresses artifacts caused during digital
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* shutdown, for example from stopping clocks.
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*/
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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snd_soc_dai_digital_mute(codec_dai, 1);
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if (cpu_dai->ops.shutdown)
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cpu_dai->ops.shutdown(substream);
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if (codec_dai->ops.shutdown)
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codec_dai->ops.shutdown(substream);
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if (machine->ops && machine->ops->shutdown)
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machine->ops->shutdown(substream);
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if (platform->pcm_ops->close)
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platform->pcm_ops->close(substream);
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cpu_dai->runtime = NULL;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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/* start delayed pop wq here for playback streams */
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codec_dai->pop_wait = 1;
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schedule_delayed_work(&socdev->delayed_work,
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msecs_to_jiffies(pmdown_time));
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} else {
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/* capture streams can be powered down now */
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snd_soc_dapm_stream_event(codec,
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codec_dai->capture.stream_name,
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SND_SOC_DAPM_STREAM_STOP);
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if (codec->active == 0 && codec_dai->pop_wait == 0)
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snd_soc_dapm_set_bias_level(socdev,
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SND_SOC_BIAS_STANDBY);
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}
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mutex_unlock(&pcm_mutex);
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return 0;
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}
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/*
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* Called by ALSA when the PCM substream is prepared, can set format, sample
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* rate, etc. This function is non atomic and can be called multiple times,
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* it can refer to the runtime info.
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*/
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static int soc_pcm_prepare(struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_device *socdev = rtd->socdev;
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struct snd_soc_dai_link *machine = rtd->dai;
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struct snd_soc_platform *platform = socdev->platform;
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struct snd_soc_dai *cpu_dai = machine->cpu_dai;
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struct snd_soc_dai *codec_dai = machine->codec_dai;
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struct snd_soc_codec *codec = socdev->codec;
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int ret = 0;
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mutex_lock(&pcm_mutex);
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if (machine->ops && machine->ops->prepare) {
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ret = machine->ops->prepare(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: machine prepare error\n");
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goto out;
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}
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}
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if (platform->pcm_ops->prepare) {
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ret = platform->pcm_ops->prepare(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: platform prepare error\n");
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goto out;
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}
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}
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if (codec_dai->ops.prepare) {
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ret = codec_dai->ops.prepare(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: codec DAI prepare error\n");
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goto out;
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}
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}
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if (cpu_dai->ops.prepare) {
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ret = cpu_dai->ops.prepare(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: cpu DAI prepare error\n");
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goto out;
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}
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}
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/* cancel any delayed stream shutdown that is pending */
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
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codec_dai->pop_wait) {
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codec_dai->pop_wait = 0;
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cancel_delayed_work(&socdev->delayed_work);
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}
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/* do we need to power up codec */
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if (codec->bias_level != SND_SOC_BIAS_ON) {
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snd_soc_dapm_set_bias_level(socdev,
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SND_SOC_BIAS_PREPARE);
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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snd_soc_dapm_stream_event(codec,
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codec_dai->playback.stream_name,
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SND_SOC_DAPM_STREAM_START);
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else
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snd_soc_dapm_stream_event(codec,
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codec_dai->capture.stream_name,
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SND_SOC_DAPM_STREAM_START);
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snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
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snd_soc_dai_digital_mute(codec_dai, 0);
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} else {
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/* codec already powered - power on widgets */
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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snd_soc_dapm_stream_event(codec,
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codec_dai->playback.stream_name,
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SND_SOC_DAPM_STREAM_START);
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else
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snd_soc_dapm_stream_event(codec,
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codec_dai->capture.stream_name,
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SND_SOC_DAPM_STREAM_START);
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snd_soc_dai_digital_mute(codec_dai, 0);
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}
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out:
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mutex_unlock(&pcm_mutex);
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return ret;
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}
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/*
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* Called by ALSA when the hardware params are set by application. This
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* function can also be called multiple times and can allocate buffers
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* (using snd_pcm_lib_* ). It's non-atomic.
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*/
|
|
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);
|
|
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);
|
|
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);
|
|
|
|
interface_err:
|
|
if (codec_dai->ops.hw_free)
|
|
codec_dai->ops.hw_free(substream);
|
|
|
|
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);
|
|
|
|
if (cpu_dai->ops.hw_free)
|
|
cpu_dai->ops.hw_free(substream);
|
|
|
|
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);
|
|
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);
|
|
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 < machine->num_links; i++) {
|
|
struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
|
|
if (dai->dai_ops.digital_mute && dai->playback.active)
|
|
dai->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->type != SND_SOC_DAI_AC97)
|
|
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->type == SND_SOC_DAI_AC97)
|
|
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_info(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->type == SND_SOC_DAI_AC97)
|
|
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 < machine->num_links; i++) {
|
|
struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
|
|
if (dai->dai_ops.digital_mute && dai->playback.active)
|
|
dai->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->type != SND_SOC_DAI_AC97)
|
|
cpu_dai->resume(pdev, cpu_dai);
|
|
if (platform->resume)
|
|
platform->resume(pdev, cpu_dai);
|
|
}
|
|
|
|
if (card->resume_post)
|
|
card->resume_post(pdev);
|
|
|
|
dev_info(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_info(socdev->dev, "scheduling resume work\n");
|
|
|
|
if (!schedule_work(&socdev->deferred_resume_work))
|
|
dev_err(socdev->dev, "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-%s-%d", dai_link->stream_name, codec_dai->name,
|
|
get_dai_name(cpu_dai->type), 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_register_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_register_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->type ==
|
|
SND_SOC_DAI_AC97_BUS)
|
|
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_register_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->type == SND_SOC_DAI_AC97_BUS && 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->dai_ops.set_sysclk)
|
|
return dai->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->dai_ops.set_clkdiv)
|
|
return dai->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->dai_ops.set_pll)
|
|
return dai->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
|
|
* @clk_id: DAI specific clock ID
|
|
* @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->dai_ops.set_fmt)
|
|
return dai->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->dai_ops.set_sysclk)
|
|
return dai->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->dai_ops.set_sysclk)
|
|
return dai->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->dai_ops.digital_mute)
|
|
return dai->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");
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MODULE_DESCRIPTION("ALSA SoC Core");
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MODULE_LICENSE("GPL");
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MODULE_ALIAS("platform:soc-audio");
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