mirror of https://gitee.com/openkylin/linux.git
1921 lines
53 KiB
C
1921 lines
53 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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* Author: Liam Girdwood
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* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.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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* Revision history
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* 12th Aug 2005 Initial version.
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* 25th Oct 2005 Working Codec, Interface and Platform registration.
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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/platform_device.h>
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#include <sound/driver.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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/* debug */
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#define SOC_DEBUG 0
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#if SOC_DEBUG
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#define dbg(format, arg...) printk(format, ## arg)
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#else
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#define dbg(format, arg...)
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#endif
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/* debug DAI capabilities matching */
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#define SOC_DEBUG_DAI 0
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#if SOC_DEBUG_DAI
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#define dbgc(format, arg...) printk(format, ## arg)
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#else
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#define dbgc(format, arg...)
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#endif
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static DEFINE_MUTEX(pcm_mutex);
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static DEFINE_MUTEX(io_mutex);
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static struct workqueue_struct *soc_workq;
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static struct work_struct soc_stream_work;
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static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
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/* supported sample rates */
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/* ATTENTION: these values depend on the definition in pcm.h! */
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static const unsigned int rates[] = {
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5512, 8000, 11025, 16000, 22050, 32000, 44100,
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48000, 64000, 88200, 96000, 176400, 192000
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};
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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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#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:
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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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/* get rate format from rate */
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static inline int soc_get_rate_format(int rate)
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{
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int i;
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for (i = 0; i < ARRAY_SIZE(rates); i++) {
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if (rates[i] == rate)
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return 1 << i;
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}
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return 0;
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}
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/* gets the audio system mclk/sysclk for the given parameters */
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static unsigned inline int soc_get_mclk(struct snd_soc_pcm_runtime *rtd,
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struct snd_soc_clock_info *info)
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{
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struct snd_soc_device *socdev = rtd->socdev;
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struct snd_soc_machine *machine = socdev->machine;
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int i;
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/* find the matching machine config and get it's mclk for the given
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* sample rate and hardware format */
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for(i = 0; i < machine->num_links; i++) {
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if (machine->dai_link[i].cpu_dai == rtd->cpu_dai &&
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machine->dai_link[i].config_sysclk)
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return machine->dai_link[i].config_sysclk(rtd, info);
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}
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return 0;
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}
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/* changes a bitclk multiplier mask to a divider mask */
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static u16 soc_bfs_mult_to_div(u16 bfs, int rate, unsigned int mclk,
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unsigned int pcmfmt, unsigned int chn)
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{
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int i, j;
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u16 bfs_ = 0;
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int size = snd_pcm_format_physical_width(pcmfmt), min = 0;
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if (size <= 0)
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return 0;
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/* the minimum bit clock that has enough bandwidth */
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min = size * rate * chn;
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dbgc("mult --> div min bclk %d with mclk %d\n", min, mclk);
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for (i = 0; i < 16; i++) {
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if ((bfs >> i) & 0x1) {
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j = rate * SND_SOC_FSB_REAL(1<<i);
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if (j >= min) {
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bfs_ |= SND_SOC_FSBD(mclk/j);
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dbgc("mult --> div support mult %d\n",
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SND_SOC_FSB_REAL(1<<i));
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}
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}
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}
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return bfs_;
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}
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/* changes a bitclk divider mask to a multiplier mask */
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static u16 soc_bfs_div_to_mult(u16 bfs, int rate, unsigned int mclk,
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unsigned int pcmfmt, unsigned int chn)
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{
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int i, j;
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u16 bfs_ = 0;
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int size = snd_pcm_format_physical_width(pcmfmt), min = 0;
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if (size <= 0)
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return 0;
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/* the minimum bit clock that has enough bandwidth */
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min = size * rate * chn;
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dbgc("div to mult min bclk %d with mclk %d\n", min, mclk);
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for (i = 0; i < 16; i++) {
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if ((bfs >> i) & 0x1) {
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j = mclk / (SND_SOC_FSBD_REAL(1<<i));
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if (j >= min) {
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bfs_ |= SND_SOC_FSB(j/rate);
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dbgc("div --> mult support div %d\n",
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SND_SOC_FSBD_REAL(1<<i));
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}
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}
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}
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return bfs_;
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}
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/* Matches codec DAI and SoC CPU DAI hardware parameters */
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static int soc_hw_match_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai_mode *codec_dai_mode = NULL;
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struct snd_soc_dai_mode *cpu_dai_mode = NULL;
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struct snd_soc_clock_info clk_info;
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unsigned int fs, mclk, codec_bfs, cpu_bfs, rate = params_rate(params),
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chn, j, k, cpu_bclk, codec_bclk, pcmrate;
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u16 fmt = 0;
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dbg("asoc: match version %s\n", SND_SOC_VERSION);
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clk_info.rate = rate;
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pcmrate = soc_get_rate_format(rate);
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/* try and find a match from the codec and cpu DAI capabilities */
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for (j = 0; j < rtd->codec_dai->caps.num_modes; j++) {
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for (k = 0; k < rtd->cpu_dai->caps.num_modes; k++) {
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codec_dai_mode = &rtd->codec_dai->caps.mode[j];
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cpu_dai_mode = &rtd->cpu_dai->caps.mode[k];
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if (!(codec_dai_mode->pcmrate & cpu_dai_mode->pcmrate &
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pcmrate)) {
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dbgc("asoc: DAI[%d:%d] failed to match rate\n", j, k);
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continue;
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}
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fmt = codec_dai_mode->fmt & cpu_dai_mode->fmt;
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if (!(fmt & SND_SOC_DAIFMT_FORMAT_MASK)) {
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dbgc("asoc: DAI[%d:%d] failed to match format\n", j, k);
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continue;
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}
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if (!(fmt & SND_SOC_DAIFMT_CLOCK_MASK)) {
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dbgc("asoc: DAI[%d:%d] failed to match clock masters\n",
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j, k);
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continue;
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}
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if (!(fmt & SND_SOC_DAIFMT_INV_MASK)) {
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dbgc("asoc: DAI[%d:%d] failed to match invert\n", j, k);
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continue;
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}
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if (!(codec_dai_mode->pcmfmt & cpu_dai_mode->pcmfmt)) {
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dbgc("asoc: DAI[%d:%d] failed to match pcm format\n", j, k);
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continue;
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}
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if (!(codec_dai_mode->pcmdir & cpu_dai_mode->pcmdir)) {
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dbgc("asoc: DAI[%d:%d] failed to match direction\n", j, k);
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continue;
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}
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/* todo - still need to add tdm selection */
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rtd->cpu_dai->dai_runtime.fmt =
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rtd->codec_dai->dai_runtime.fmt =
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1 << (ffs(fmt & SND_SOC_DAIFMT_FORMAT_MASK) -1) |
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1 << (ffs(fmt & SND_SOC_DAIFMT_CLOCK_MASK) - 1) |
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1 << (ffs(fmt & SND_SOC_DAIFMT_INV_MASK) - 1);
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clk_info.bclk_master =
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rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK;
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/* make sure the ratio between rate and master
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* clock is acceptable*/
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fs = (cpu_dai_mode->fs & codec_dai_mode->fs);
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if (fs == 0) {
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dbgc("asoc: DAI[%d:%d] failed to match FS\n", j, k);
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continue;
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}
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clk_info.fs = rtd->cpu_dai->dai_runtime.fs =
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rtd->codec_dai->dai_runtime.fs = fs;
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/* calculate audio system clocking using slowest clocks possible*/
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mclk = soc_get_mclk(rtd, &clk_info);
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if (mclk == 0) {
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dbgc("asoc: DAI[%d:%d] configuration not clockable\n", j, k);
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dbgc("asoc: rate %d fs %d master %x\n", rate, fs,
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clk_info.bclk_master);
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continue;
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}
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/* calculate word size (per channel) and frame size */
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rtd->codec_dai->dai_runtime.pcmfmt =
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rtd->cpu_dai->dai_runtime.pcmfmt =
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1 << params_format(params);
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chn = params_channels(params);
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/* i2s always has left and right */
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if (params_channels(params) == 1 &&
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rtd->cpu_dai->dai_runtime.fmt & (SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_LEFT_J))
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chn <<= 1;
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/* Calculate bfs - the ratio between bitclock and the sample rate
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* We must take into consideration the dividers and multipliers
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* used in the codec and cpu DAI modes. We always choose the
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* lowest possible clocks to reduce power.
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*/
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if (codec_dai_mode->flags & cpu_dai_mode->flags &
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SND_SOC_DAI_BFS_DIV) {
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/* cpu & codec bfs dividers */
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rtd->cpu_dai->dai_runtime.bfs =
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rtd->codec_dai->dai_runtime.bfs =
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1 << (fls(codec_dai_mode->bfs & cpu_dai_mode->bfs) - 1);
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} else if (codec_dai_mode->flags & SND_SOC_DAI_BFS_DIV) {
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/* normalise bfs codec divider & cpu mult */
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codec_bfs = soc_bfs_div_to_mult(codec_dai_mode->bfs, rate,
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mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn);
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rtd->cpu_dai->dai_runtime.bfs =
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1 << (ffs(codec_bfs & cpu_dai_mode->bfs) - 1);
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cpu_bfs = soc_bfs_mult_to_div(cpu_dai_mode->bfs, rate, mclk,
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rtd->codec_dai->dai_runtime.pcmfmt, chn);
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rtd->codec_dai->dai_runtime.bfs =
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1 << (fls(codec_dai_mode->bfs & cpu_bfs) - 1);
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} else if (cpu_dai_mode->flags & SND_SOC_DAI_BFS_DIV) {
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/* normalise bfs codec mult & cpu divider */
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codec_bfs = soc_bfs_mult_to_div(codec_dai_mode->bfs, rate,
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mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn);
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rtd->cpu_dai->dai_runtime.bfs =
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1 << (fls(codec_bfs & cpu_dai_mode->bfs) -1);
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cpu_bfs = soc_bfs_div_to_mult(cpu_dai_mode->bfs, rate, mclk,
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rtd->codec_dai->dai_runtime.pcmfmt, chn);
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rtd->codec_dai->dai_runtime.bfs =
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1 << (ffs(codec_dai_mode->bfs & cpu_bfs) -1);
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} else {
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/* codec & cpu bfs rate multipliers */
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rtd->cpu_dai->dai_runtime.bfs =
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rtd->codec_dai->dai_runtime.bfs =
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1 << (ffs(codec_dai_mode->bfs & cpu_dai_mode->bfs) -1);
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}
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/* make sure the bit clock speed is acceptable */
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if (!rtd->cpu_dai->dai_runtime.bfs ||
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!rtd->codec_dai->dai_runtime.bfs) {
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dbgc("asoc: DAI[%d:%d] failed to match BFS\n", j, k);
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dbgc("asoc: cpu_dai %x codec %x\n",
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rtd->cpu_dai->dai_runtime.bfs,
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rtd->codec_dai->dai_runtime.bfs);
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dbgc("asoc: mclk %d hwfmt %x\n", mclk, fmt);
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continue;
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}
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goto found;
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}
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}
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printk(KERN_ERR "asoc: no matching DAI found between codec and CPU\n");
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return -EINVAL;
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found:
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/* we have matching DAI's, so complete the runtime info */
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rtd->codec_dai->dai_runtime.pcmrate =
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rtd->cpu_dai->dai_runtime.pcmrate =
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soc_get_rate_format(rate);
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rtd->codec_dai->dai_runtime.priv = codec_dai_mode->priv;
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rtd->cpu_dai->dai_runtime.priv = cpu_dai_mode->priv;
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rtd->codec_dai->dai_runtime.flags = codec_dai_mode->flags;
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rtd->cpu_dai->dai_runtime.flags = cpu_dai_mode->flags;
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/* for debug atm */
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dbg("asoc: DAI[%d:%d] Match OK\n", j, k);
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if (rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) {
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codec_bclk = (rtd->codec_dai->dai_runtime.fs * params_rate(params)) /
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SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs);
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dbg("asoc: codec fs %d mclk %d bfs div %d bclk %d\n",
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rtd->codec_dai->dai_runtime.fs, mclk,
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SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk);
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} else {
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codec_bclk = params_rate(params) *
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SND_SOC_FSB_REAL(rtd->codec_dai->dai_runtime.bfs);
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dbg("asoc: codec fs %d mclk %d bfs mult %d bclk %d\n",
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rtd->codec_dai->dai_runtime.fs, mclk,
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SND_SOC_FSB_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk);
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}
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if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) {
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cpu_bclk = (rtd->cpu_dai->dai_runtime.fs * params_rate(params)) /
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SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs);
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dbg("asoc: cpu fs %d mclk %d bfs div %d bclk %d\n",
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rtd->cpu_dai->dai_runtime.fs, mclk,
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SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk);
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} else {
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cpu_bclk = params_rate(params) *
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SND_SOC_FSB_REAL(rtd->cpu_dai->dai_runtime.bfs);
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dbg("asoc: cpu fs %d mclk %d bfs mult %d bclk %d\n",
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rtd->cpu_dai->dai_runtime.fs, mclk,
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SND_SOC_FSB_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk);
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}
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/*
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* Check we have matching bitclocks. If we don't then it means the
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* sysclock returned by either the codec or cpu DAI (selected by the
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* machine sysclock function) is wrong compared with the supported DAI
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* modes for the codec or cpu DAI.
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*/
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if (cpu_bclk != codec_bclk){
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printk(KERN_ERR
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"asoc: codec and cpu bitclocks differ, audio may be wrong speed\n"
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);
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printk(KERN_ERR "asoc: codec %d != cpu %d\n", codec_bclk, cpu_bclk);
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}
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switch(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
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case SND_SOC_DAIFMT_CBM_CFM:
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dbg("asoc: DAI codec BCLK master, LRC master\n");
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break;
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case SND_SOC_DAIFMT_CBS_CFM:
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dbg("asoc: DAI codec BCLK slave, LRC master\n");
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break;
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case SND_SOC_DAIFMT_CBM_CFS:
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dbg("asoc: DAI codec BCLK master, LRC slave\n");
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break;
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case SND_SOC_DAIFMT_CBS_CFS:
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dbg("asoc: DAI codec BCLK slave, LRC slave\n");
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break;
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}
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dbg("asoc: mode %x, invert %x\n",
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rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK,
|
|
rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK);
|
|
dbg("asoc: audio rate %d chn %d fmt %x\n", params_rate(params),
|
|
params_channels(params), params_format(params));
|
|
|
|
return 0;
|
|
}
|
|
|
|
static inline u32 get_rates(struct snd_soc_dai_mode *modes, int nmodes)
|
|
{
|
|
int i;
|
|
u32 rates = 0;
|
|
|
|
for(i = 0; i < nmodes; i++)
|
|
rates |= modes[i].pcmrate;
|
|
|
|
return rates;
|
|
}
|
|
|
|
static inline u64 get_formats(struct snd_soc_dai_mode *modes, int nmodes)
|
|
{
|
|
int i;
|
|
u64 formats = 0;
|
|
|
|
for(i = 0; i < nmodes; i++)
|
|
formats |= modes[i].pcmfmt;
|
|
|
|
return formats;
|
|
}
|
|
|
|
/*
|
|
* 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_machine *machine = socdev->machine;
|
|
struct snd_soc_platform *platform = socdev->platform;
|
|
struct snd_soc_codec_dai *codec_dai = rtd->codec_dai;
|
|
struct snd_soc_cpu_dai *cpu_dai = rtd->cpu_dai;
|
|
int ret = 0;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
|
|
/* startup the audio subsystem */
|
|
if (rtd->cpu_dai->ops.startup) {
|
|
ret = rtd->cpu_dai->ops.startup(substream);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't open interface %s\n",
|
|
rtd->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 (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;
|
|
}
|
|
}
|
|
|
|
if (rtd->codec_dai->ops.startup) {
|
|
ret = rtd->codec_dai->ops.startup(substream);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't open codec %s\n",
|
|
rtd->codec_dai->name);
|
|
goto codec_dai_err;
|
|
}
|
|
}
|
|
|
|
/* create runtime params from DMA, codec and cpu DAI */
|
|
if (runtime->hw.rates)
|
|
runtime->hw.rates &=
|
|
get_rates(codec_dai->caps.mode, codec_dai->caps.num_modes) &
|
|
get_rates(cpu_dai->caps.mode, cpu_dai->caps.num_modes);
|
|
else
|
|
runtime->hw.rates =
|
|
get_rates(codec_dai->caps.mode, codec_dai->caps.num_modes) &
|
|
get_rates(cpu_dai->caps.mode, cpu_dai->caps.num_modes);
|
|
if (runtime->hw.formats)
|
|
runtime->hw.formats &=
|
|
get_formats(codec_dai->caps.mode, codec_dai->caps.num_modes) &
|
|
get_formats(cpu_dai->caps.mode, cpu_dai->caps.num_modes);
|
|
else
|
|
runtime->hw.formats =
|
|
get_formats(codec_dai->caps.mode, codec_dai->caps.num_modes) &
|
|
get_formats(cpu_dai->caps.mode, cpu_dai->caps.num_modes);
|
|
|
|
/* Check that the codec and cpu DAI's are compatible */
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
|
runtime->hw.rate_min =
|
|
max(rtd->codec_dai->playback.rate_min,
|
|
rtd->cpu_dai->playback.rate_min);
|
|
runtime->hw.rate_max =
|
|
min(rtd->codec_dai->playback.rate_max,
|
|
rtd->cpu_dai->playback.rate_max);
|
|
runtime->hw.channels_min =
|
|
max(rtd->codec_dai->playback.channels_min,
|
|
rtd->cpu_dai->playback.channels_min);
|
|
runtime->hw.channels_max =
|
|
min(rtd->codec_dai->playback.channels_max,
|
|
rtd->cpu_dai->playback.channels_max);
|
|
} else {
|
|
runtime->hw.rate_min =
|
|
max(rtd->codec_dai->capture.rate_min,
|
|
rtd->cpu_dai->capture.rate_min);
|
|
runtime->hw.rate_max =
|
|
min(rtd->codec_dai->capture.rate_max,
|
|
rtd->cpu_dai->capture.rate_max);
|
|
runtime->hw.channels_min =
|
|
max(rtd->codec_dai->capture.channels_min,
|
|
rtd->cpu_dai->capture.channels_min);
|
|
runtime->hw.channels_max =
|
|
min(rtd->codec_dai->capture.channels_max,
|
|
rtd->cpu_dai->capture.channels_max);
|
|
}
|
|
|
|
snd_pcm_limit_hw_rates(runtime);
|
|
if (!runtime->hw.rates) {
|
|
printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
|
|
rtd->codec_dai->name, rtd->cpu_dai->name);
|
|
goto codec_dai_err;
|
|
}
|
|
if (!runtime->hw.formats) {
|
|
printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
|
|
rtd->codec_dai->name, rtd->cpu_dai->name);
|
|
goto codec_dai_err;
|
|
}
|
|
if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
|
|
printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
|
|
rtd->codec_dai->name, rtd->cpu_dai->name);
|
|
goto codec_dai_err;
|
|
}
|
|
|
|
dbg("asoc: %s <-> %s info:\n", rtd->codec_dai->name, rtd->cpu_dai->name);
|
|
dbg("asoc: rate mask 0x%x \nasoc: min ch %d max ch %d\n"
|
|
"asoc: min rate %d max rate %d\n",
|
|
runtime->hw.rates, runtime->hw.channels_min,
|
|
runtime->hw.channels_max, runtime->hw.rate_min, runtime->hw.rate_max);
|
|
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
rtd->cpu_dai->playback.active = rtd->codec_dai->playback.active = 1;
|
|
else
|
|
rtd->cpu_dai->capture.active = rtd->codec_dai->capture.active = 1;
|
|
rtd->cpu_dai->active = rtd->codec_dai->active = 1;
|
|
rtd->cpu_dai->runtime = runtime;
|
|
socdev->codec->active++;
|
|
mutex_unlock(&pcm_mutex);
|
|
return 0;
|
|
|
|
codec_dai_err:
|
|
if (machine->ops && machine->ops->shutdown)
|
|
machine->ops->shutdown(substream);
|
|
|
|
machine_err:
|
|
if (platform->pcm_ops->close)
|
|
platform->pcm_ops->close(substream);
|
|
|
|
platform_err:
|
|
if (rtd->cpu_dai->ops.shutdown)
|
|
rtd->cpu_dai->ops.shutdown(substream);
|
|
out:
|
|
mutex_unlock(&pcm_mutex);
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* Power down the audio subsytem 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(void *data)
|
|
{
|
|
struct snd_soc_device *socdev = data;
|
|
struct snd_soc_codec *codec = socdev->codec;
|
|
struct snd_soc_codec_dai *codec_dai;
|
|
int i;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
for(i = 0; i < codec->num_dai; i++) {
|
|
codec_dai = &codec->dai[i];
|
|
|
|
dbg("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) {
|
|
|
|
codec_dai->pop_wait = 0;
|
|
snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name,
|
|
SND_SOC_DAPM_STREAM_STOP);
|
|
|
|
/* power down the codec power domain if no longer active */
|
|
if (codec->active == 0) {
|
|
dbg("pop wq D3 %s %s\n", codec->name,
|
|
codec_dai->playback.stream_name);
|
|
if (codec->dapm_event)
|
|
codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
|
|
}
|
|
}
|
|
}
|
|
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_machine *machine = socdev->machine;
|
|
struct snd_soc_platform *platform = socdev->platform;
|
|
struct snd_soc_codec *codec = socdev->codec;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
rtd->cpu_dai->playback.active = rtd->codec_dai->playback.active = 0;
|
|
else
|
|
rtd->cpu_dai->capture.active = rtd->codec_dai->capture.active = 0;
|
|
|
|
if (rtd->codec_dai->playback.active == 0 &&
|
|
rtd->codec_dai->capture.active == 0) {
|
|
rtd->cpu_dai->active = rtd->codec_dai->active = 0;
|
|
}
|
|
codec->active--;
|
|
|
|
if (rtd->cpu_dai->ops.shutdown)
|
|
rtd->cpu_dai->ops.shutdown(substream);
|
|
|
|
if (rtd->codec_dai->ops.shutdown)
|
|
rtd->codec_dai->ops.shutdown(substream);
|
|
|
|
if (machine->ops && machine->ops->shutdown)
|
|
machine->ops->shutdown(substream);
|
|
|
|
if (platform->pcm_ops->close)
|
|
platform->pcm_ops->close(substream);
|
|
rtd->cpu_dai->runtime = NULL;
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
|
/* start delayed pop wq here for playback streams */
|
|
rtd->codec_dai->pop_wait = 1;
|
|
queue_delayed_work(soc_workq, &soc_stream_work,
|
|
msecs_to_jiffies(pmdown_time));
|
|
} else {
|
|
/* capture streams can be powered down now */
|
|
snd_soc_dapm_stream_event(codec, rtd->codec_dai->capture.stream_name,
|
|
SND_SOC_DAPM_STREAM_STOP);
|
|
|
|
if (codec->active == 0 && rtd->codec_dai->pop_wait == 0){
|
|
if (codec->dapm_event)
|
|
codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
|
|
}
|
|
}
|
|
|
|
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_platform *platform = socdev->platform;
|
|
struct snd_soc_codec *codec = socdev->codec;
|
|
int ret = 0;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
if (platform->pcm_ops->prepare) {
|
|
ret = platform->pcm_ops->prepare(substream);
|
|
if (ret < 0)
|
|
goto out;
|
|
}
|
|
|
|
if (rtd->codec_dai->ops.prepare) {
|
|
ret = rtd->codec_dai->ops.prepare(substream);
|
|
if (ret < 0)
|
|
goto out;
|
|
}
|
|
|
|
if (rtd->cpu_dai->ops.prepare)
|
|
ret = rtd->cpu_dai->ops.prepare(substream);
|
|
|
|
/* we only want to start a DAPM playback stream if we are not waiting
|
|
* on an existing one stopping */
|
|
if (rtd->codec_dai->pop_wait) {
|
|
/* we are waiting for the delayed work to start */
|
|
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
|
|
snd_soc_dapm_stream_event(codec,
|
|
rtd->codec_dai->capture.stream_name,
|
|
SND_SOC_DAPM_STREAM_START);
|
|
else {
|
|
rtd->codec_dai->pop_wait = 0;
|
|
cancel_delayed_work(&soc_stream_work);
|
|
if (rtd->codec_dai->digital_mute)
|
|
rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0);
|
|
}
|
|
} else {
|
|
/* no delayed work - do we need to power up codec */
|
|
if (codec->dapm_state != SNDRV_CTL_POWER_D0) {
|
|
|
|
if (codec->dapm_event)
|
|
codec->dapm_event(codec, SNDRV_CTL_POWER_D1);
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
snd_soc_dapm_stream_event(codec,
|
|
rtd->codec_dai->playback.stream_name,
|
|
SND_SOC_DAPM_STREAM_START);
|
|
else
|
|
snd_soc_dapm_stream_event(codec,
|
|
rtd->codec_dai->capture.stream_name,
|
|
SND_SOC_DAPM_STREAM_START);
|
|
|
|
if (codec->dapm_event)
|
|
codec->dapm_event(codec, SNDRV_CTL_POWER_D0);
|
|
if (rtd->codec_dai->digital_mute)
|
|
rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0);
|
|
|
|
} else {
|
|
/* codec already powered - power on widgets */
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
snd_soc_dapm_stream_event(codec,
|
|
rtd->codec_dai->playback.stream_name,
|
|
SND_SOC_DAPM_STREAM_START);
|
|
else
|
|
snd_soc_dapm_stream_event(codec,
|
|
rtd->codec_dai->capture.stream_name,
|
|
SND_SOC_DAPM_STREAM_START);
|
|
if (rtd->codec_dai->digital_mute)
|
|
rtd->codec_dai->digital_mute(codec, rtd->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_platform *platform = socdev->platform;
|
|
struct snd_soc_machine *machine = socdev->machine;
|
|
int ret = 0;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
|
|
/* we don't need to match any AC97 params */
|
|
if (rtd->cpu_dai->type != SND_SOC_DAI_AC97) {
|
|
ret = soc_hw_match_params(substream, params);
|
|
if (ret < 0)
|
|
goto out;
|
|
} else {
|
|
struct snd_soc_clock_info clk_info;
|
|
clk_info.rate = params_rate(params);
|
|
ret = soc_get_mclk(rtd, &clk_info);
|
|
if (ret < 0)
|
|
goto out;
|
|
}
|
|
|
|
if (rtd->codec_dai->ops.hw_params) {
|
|
ret = rtd->codec_dai->ops.hw_params(substream, params);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't set codec %s hw params\n",
|
|
rtd->codec_dai->name);
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
if (rtd->cpu_dai->ops.hw_params) {
|
|
ret = rtd->cpu_dai->ops.hw_params(substream, params);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't set interface %s hw params\n",
|
|
rtd->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: can't set platform %s hw params\n",
|
|
platform->name);
|
|
goto platform_err;
|
|
}
|
|
}
|
|
|
|
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 machine_err;
|
|
}
|
|
}
|
|
|
|
out:
|
|
mutex_unlock(&pcm_mutex);
|
|
return ret;
|
|
|
|
machine_err:
|
|
if (platform->pcm_ops->hw_free)
|
|
platform->pcm_ops->hw_free(substream);
|
|
|
|
platform_err:
|
|
if (rtd->cpu_dai->ops.hw_free)
|
|
rtd->cpu_dai->ops.hw_free(substream);
|
|
|
|
interface_err:
|
|
if (rtd->codec_dai->ops.hw_free)
|
|
rtd->codec_dai->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_platform *platform = socdev->platform;
|
|
struct snd_soc_codec *codec = socdev->codec;
|
|
struct snd_soc_machine *machine = socdev->machine;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
|
|
/* apply codec digital mute */
|
|
if (!codec->active && rtd->codec_dai->digital_mute)
|
|
rtd->codec_dai->digital_mute(codec, rtd->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 (rtd->codec_dai->ops.hw_free)
|
|
rtd->codec_dai->ops.hw_free(substream);
|
|
|
|
if (rtd->cpu_dai->ops.hw_free)
|
|
rtd->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_platform *platform = socdev->platform;
|
|
int ret;
|
|
|
|
if (rtd->codec_dai->ops.trigger) {
|
|
ret = rtd->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 (rtd->cpu_dai->ops.trigger) {
|
|
ret = rtd->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_machine *machine = socdev->machine;
|
|
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;
|
|
|
|
/* mute any active DAC's */
|
|
for(i = 0; i < machine->num_links; i++) {
|
|
struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
|
|
if (dai->digital_mute && dai->playback.active)
|
|
dai->digital_mute(codec, dai, 1);
|
|
}
|
|
|
|
if (machine->suspend_pre)
|
|
machine->suspend_pre(pdev, state);
|
|
|
|
for(i = 0; i < machine->num_links; i++) {
|
|
struct snd_soc_cpu_dai *cpu_dai = machine->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 */
|
|
flush_workqueue(soc_workq);
|
|
codec->suspend_dapm_state = codec->dapm_state;
|
|
|
|
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 < machine->num_links; i++) {
|
|
struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
|
|
if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
|
|
cpu_dai->suspend(pdev, cpu_dai);
|
|
}
|
|
|
|
if (machine->suspend_post)
|
|
machine->suspend_post(pdev, state);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* powers up audio subsystem after a suspend */
|
|
static int soc_resume(struct platform_device *pdev)
|
|
{
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_machine *machine = socdev->machine;
|
|
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;
|
|
|
|
if (machine->resume_pre)
|
|
machine->resume_pre(pdev);
|
|
|
|
for(i = 0; i < machine->num_links; i++) {
|
|
struct snd_soc_cpu_dai *cpu_dai = machine->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 DAC's */
|
|
for(i = 0; i < machine->num_links; i++) {
|
|
struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
|
|
if (dai->digital_mute && dai->playback.active)
|
|
dai->digital_mute(codec, dai, 0);
|
|
}
|
|
|
|
for(i = 0; i < machine->num_links; i++) {
|
|
struct snd_soc_cpu_dai *cpu_dai = machine->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 (machine->resume_post)
|
|
machine->resume_post(pdev);
|
|
|
|
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_machine *machine = socdev->machine;
|
|
struct snd_soc_platform *platform = socdev->platform;
|
|
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
|
|
|
|
if (machine->probe) {
|
|
ret = machine->probe(pdev);
|
|
if(ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
for (i = 0; i < machine->num_links; i++) {
|
|
struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
|
|
if (cpu_dai->probe) {
|
|
ret = cpu_dai->probe(pdev);
|
|
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 */
|
|
soc_workq = create_workqueue("kdapm");
|
|
if (soc_workq == NULL)
|
|
goto work_err;
|
|
INIT_WORK(&soc_stream_work, close_delayed_work, socdev);
|
|
return 0;
|
|
|
|
work_err:
|
|
if (platform->remove)
|
|
platform->remove(pdev);
|
|
|
|
platform_err:
|
|
if (codec_dev->remove)
|
|
codec_dev->remove(pdev);
|
|
|
|
cpu_dai_err:
|
|
for (i--; i > 0; i--) {
|
|
struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
|
|
if (cpu_dai->remove)
|
|
cpu_dai->remove(pdev);
|
|
}
|
|
|
|
if (machine->remove)
|
|
machine->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_machine *machine = socdev->machine;
|
|
struct snd_soc_platform *platform = socdev->platform;
|
|
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
|
|
|
|
if (soc_workq)
|
|
destroy_workqueue(soc_workq);
|
|
|
|
if (platform->remove)
|
|
platform->remove(pdev);
|
|
|
|
if (codec_dev->remove)
|
|
codec_dev->remove(pdev);
|
|
|
|
for (i = 0; i < machine->num_links; i++) {
|
|
struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
|
|
if (cpu_dai->remove)
|
|
cpu_dai->remove(pdev);
|
|
}
|
|
|
|
if (machine->remove)
|
|
machine->remove(pdev);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* ASoC platform driver */
|
|
static struct platform_driver soc_driver = {
|
|
.driver = {
|
|
.name = "soc-audio",
|
|
},
|
|
.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_codec_dai *codec_dai = dai_link->codec_dai;
|
|
struct snd_soc_cpu_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->cpu_dai = cpu_dai;
|
|
rtd->codec_dai = codec_dai;
|
|
rtd->socdev = socdev;
|
|
|
|
/* 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;
|
|
}
|
|
|
|
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 codec_reg_show(struct device *dev,
|
|
struct device_attribute *attr, char *buf)
|
|
{
|
|
struct snd_soc_device *devdata = dev_get_drvdata(dev);
|
|
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: %4x\n", i, codec->read(codec, i));
|
|
|
|
return count;
|
|
}
|
|
static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
|
|
|
|
/**
|
|
* 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_get_rate - get int sample rate
|
|
* @hwpcmrate: the hardware pcm rate
|
|
*
|
|
* Returns the audio rate integaer value, else 0.
|
|
*/
|
|
int snd_soc_get_rate(int hwpcmrate)
|
|
{
|
|
int rate = ffs(hwpcmrate) - 1;
|
|
|
|
if (rate > ARRAY_SIZE(rates))
|
|
return 0;
|
|
return rates[rate];
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_get_rate);
|
|
|
|
/**
|
|
* 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_machine *machine = socdev->machine;
|
|
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 < machine->num_links; i++) {
|
|
ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't create pcm %s\n",
|
|
machine->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_machine *machine = socdev->machine;
|
|
int ret = 0, i, ac97 = 0;
|
|
|
|
mutex_lock(&codec->mutex);
|
|
for(i = 0; i < machine->num_links; i++) {
|
|
if (socdev->machine->dai_link[i].init)
|
|
socdev->machine->dai_link[i].init(codec);
|
|
if (socdev->machine->dai_link[i].cpu_dai->type == SND_SOC_DAI_AC97)
|
|
ac97 = 1;
|
|
}
|
|
snprintf(codec->card->shortname, sizeof(codec->card->shortname),
|
|
"%s", machine->name);
|
|
snprintf(codec->card->longname, sizeof(codec->card->longname),
|
|
"%s (%s)", machine->name, codec->name);
|
|
|
|
ret = snd_card_register(codec->card);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n",
|
|
codec->name);
|
|
mutex_unlock(&codec->mutex);
|
|
return ret;
|
|
}
|
|
|
|
#ifdef CONFIG_SND_SOC_AC97_BUS
|
|
if (ac97)
|
|
soc_ac97_dev_register(codec);
|
|
#endif
|
|
|
|
snd_soc_dapm_sys_add(socdev->dev);
|
|
device_create_file(socdev->dev, &dev_attr_codec_reg);
|
|
mutex_unlock(&codec->mutex);
|
|
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;
|
|
|
|
mutex_lock(&codec->mutex);
|
|
#ifdef CONFIG_SND_SOC_AC97_BUS
|
|
if (codec->ac97)
|
|
soc_ac97_dev_unregister(codec);
|
|
#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.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
|
|
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->mask;
|
|
|
|
if (uinfo->value.enumerated.item > e->mask - 1)
|
|
uinfo->value.enumerated.item = e->mask - 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->mask; 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->mask; bitmask <<= 1)
|
|
;
|
|
if (ucontrol->value.enumerated.item[0] > e->mask - 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->mask - 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->mask;
|
|
|
|
if (uinfo->value.enumerated.item > e->mask - 1)
|
|
uinfo->value.enumerated.item = e->mask - 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 mask = kcontrol->private_value;
|
|
|
|
uinfo->type =
|
|
mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
|
|
uinfo->count = 1;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = mask;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
|
|
|
|
/**
|
|
* snd_soc_info_bool_ext - external single boolean mixer info callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Callback to provide information about a single boolean external mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_info *uinfo)
|
|
{
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
|
|
uinfo->count = 1;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = 1;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_bool_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)
|
|
{
|
|
int mask = (kcontrol->private_value >> 16) & 0xff;
|
|
int shift = (kcontrol->private_value >> 8) & 0x0f;
|
|
int rshift = (kcontrol->private_value >> 12) & 0x0f;
|
|
|
|
uinfo->type =
|
|
mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
|
|
uinfo->count = shift == rshift ? 1 : 2;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = mask;
|
|
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 snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
int reg = kcontrol->private_value & 0xff;
|
|
int shift = (kcontrol->private_value >> 8) & 0x0f;
|
|
int rshift = (kcontrol->private_value >> 12) & 0x0f;
|
|
int mask = (kcontrol->private_value >> 16) & 0xff;
|
|
int invert = (kcontrol->private_value >> 24) & 0x01;
|
|
|
|
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] =
|
|
mask - ucontrol->value.integer.value[0];
|
|
if (shift != rshift)
|
|
ucontrol->value.integer.value[1] =
|
|
mask - 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 snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
int reg = kcontrol->private_value & 0xff;
|
|
int shift = (kcontrol->private_value >> 8) & 0x0f;
|
|
int rshift = (kcontrol->private_value >> 12) & 0x0f;
|
|
int mask = (kcontrol->private_value >> 16) & 0xff;
|
|
int invert = (kcontrol->private_value >> 24) & 0x01;
|
|
int err;
|
|
unsigned short val, val2, val_mask;
|
|
|
|
val = (ucontrol->value.integer.value[0] & mask);
|
|
if (invert)
|
|
val = mask - val;
|
|
val_mask = mask << shift;
|
|
val = val << shift;
|
|
if (shift != rshift) {
|
|
val2 = (ucontrol->value.integer.value[1] & mask);
|
|
if (invert)
|
|
val2 = mask - val2;
|
|
val_mask |= mask << rshift;
|
|
val |= val2 << rshift;
|
|
}
|
|
err = snd_soc_update_bits(codec, reg, val_mask, val);
|
|
return err;
|
|
}
|
|
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)
|
|
{
|
|
int mask = (kcontrol->private_value >> 12) & 0xff;
|
|
|
|
uinfo->type =
|
|
mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
|
|
uinfo->count = 2;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = mask;
|
|
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 snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
int reg = kcontrol->private_value & 0xff;
|
|
int reg2 = (kcontrol->private_value >> 24) & 0xff;
|
|
int shift = (kcontrol->private_value >> 8) & 0x0f;
|
|
int mask = (kcontrol->private_value >> 12) & 0xff;
|
|
int invert = (kcontrol->private_value >> 20) & 0x01;
|
|
|
|
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] =
|
|
mask - ucontrol->value.integer.value[0];
|
|
ucontrol->value.integer.value[1] =
|
|
mask - 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 snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
int reg = kcontrol->private_value & 0xff;
|
|
int reg2 = (kcontrol->private_value >> 24) & 0xff;
|
|
int shift = (kcontrol->private_value >> 8) & 0x0f;
|
|
int mask = (kcontrol->private_value >> 12) & 0xff;
|
|
int invert = (kcontrol->private_value >> 20) & 0x01;
|
|
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 = mask - val;
|
|
val2 = mask - val2;
|
|
}
|
|
|
|
val = val << shift;
|
|
val2 = val2 << shift;
|
|
|
|
if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0)
|
|
return err;
|
|
|
|
err = snd_soc_update_bits(codec, reg2, val_mask, val2);
|
|
return err;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
|
|
|
|
static int __devinit snd_soc_init(void)
|
|
{
|
|
printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
|
|
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, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
|
|
MODULE_DESCRIPTION("ALSA SoC Core");
|
|
MODULE_LICENSE("GPL");
|