/* * soc-core.c -- ALSA SoC Audio Layer * * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * Author: Liam Girdwood * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com * with code, comments and ideas from :- * Richard Purdie * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * * Revision history * 12th Aug 2005 Initial version. * 25th Oct 2005 Working Codec, Interface and Platform registration. * * TODO: * o Add hw rules to enforce rates, etc. * o More testing with other codecs/machines. * o Add more codecs and platforms to ensure good API coverage. * o Support TDM on PCM and I2S */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include /* debug */ #define SOC_DEBUG 0 #if SOC_DEBUG #define dbg(format, arg...) printk(format, ## arg) #else #define dbg(format, arg...) #endif /* debug DAI capabilities matching */ #define SOC_DEBUG_DAI 0 #if SOC_DEBUG_DAI #define dbgc(format, arg...) printk(format, ## arg) #else #define dbgc(format, arg...) #endif #define CODEC_CPU(codec, cpu) ((codec << 4) | cpu) static DEFINE_MUTEX(pcm_mutex); static DEFINE_MUTEX(io_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); /* supported sample rates */ /* ATTENTION: these values depend on the definition in pcm.h! */ static const unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000 }; /* * This is a timeout to do a DAPM powerdown after a stream is closed(). * It can be used to eliminate pops between different playback streams, e.g. * between two audio tracks. */ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); #ifdef CONFIG_SND_SOC_AC97_BUS /* unregister ac97 codec */ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) { if (codec->ac97->dev.bus) device_unregister(&codec->ac97->dev); return 0; } /* stop no dev release warning */ static void soc_ac97_device_release(struct device *dev){} /* register ac97 codec to bus */ static int soc_ac97_dev_register(struct snd_soc_codec *codec) { int err; codec->ac97->dev.bus = &ac97_bus_type; codec->ac97->dev.parent = NULL; codec->ac97->dev.release = soc_ac97_device_release; snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", codec->card->number, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); codec->ac97->dev.bus = NULL; return err; } return 0; } #endif static inline const char* get_dai_name(int type) { switch(type) { case SND_SOC_DAI_AC97: return "AC97"; case SND_SOC_DAI_I2S: return "I2S"; case SND_SOC_DAI_PCM: return "PCM"; } return NULL; } /* get rate format from rate */ static inline int soc_get_rate_format(int rate) { int i; for (i = 0; i < ARRAY_SIZE(rates); i++) { if (rates[i] == rate) return 1 << i; } return 0; } /* gets the audio system mclk/sysclk for the given parameters */ static unsigned inline int soc_get_mclk(struct snd_soc_pcm_runtime *rtd, struct snd_soc_clock_info *info) { struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_machine *machine = socdev->machine; int i; /* find the matching machine config and get it's mclk for the given * sample rate and hardware format */ for(i = 0; i < machine->num_links; i++) { if (machine->dai_link[i].cpu_dai == rtd->cpu_dai && machine->dai_link[i].config_sysclk) return machine->dai_link[i].config_sysclk(rtd, info); } return 0; } /* changes a bitclk multiplier mask to a divider mask */ static u64 soc_bfs_rcw_to_div(u64 bfs, int rate, unsigned int mclk, unsigned int pcmfmt, unsigned int chn) { int i, j; u64 bfs_ = 0; int size = snd_pcm_format_physical_width(pcmfmt), min = 0; if (size <= 0) return 0; /* the minimum bit clock that has enough bandwidth */ min = size * rate * chn; dbgc("rcw --> div min bclk %d with mclk %d\n", min, mclk); for (i = 0; i < 64; i++) { if ((bfs >> i) & 0x1) { j = min * (i + 1); bfs_ |= SND_SOC_FSBD(mclk/j); dbgc("rcw --> div support mult %d\n", SND_SOC_FSBD_REAL(1<> i) & 0x1) { j = mclk / (i + 1); if (j >= min) { bfs_ |= SND_SOC_FSBW(j/min); dbgc("div --> rcw support div %d\n", SND_SOC_FSBW_REAL(1< rcw min bclk %d with mclk %d\n", min, mclk); if (bfs_ < min) return 0; else { bfs_ = SND_SOC_FSBW(bfs_/min); dbgc("rate --> rcw support div %d\n", SND_SOC_FSBW_REAL(bfs_)); return bfs_; } } /* changes a bitclk multiplier mask to a divider mask */ static u64 soc_bfs_rate_to_div(u64 bfs, int rate, unsigned int mclk, unsigned int pcmfmt, unsigned int chn) { unsigned int bfs_ = rate * bfs; int size = snd_pcm_format_physical_width(pcmfmt), min = 0; if (size <= 0) return 0; /* the minimum bit clock that has enough bandwidth */ min = size * rate * chn; dbgc("rate --> div min bclk %d with mclk %d\n", min, mclk); if (bfs_ < min) return 0; else { bfs_ = SND_SOC_FSBW(mclk/bfs_); dbgc("rate --> div support div %d\n", SND_SOC_FSBD_REAL(bfs_)); return bfs_; } } /* Matches codec DAI and SoC CPU DAI hardware parameters */ static int soc_hw_match_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_mode *codec_dai_mode = NULL; struct snd_soc_dai_mode *cpu_dai_mode = NULL; struct snd_soc_clock_info clk_info; unsigned int fs, mclk, rate = params_rate(params), chn, j, k, cpu_bclk, codec_bclk, pcmrate; u16 fmt = 0; u64 codec_bfs, cpu_bfs; dbg("asoc: match version %s\n", SND_SOC_VERSION); clk_info.rate = rate; pcmrate = soc_get_rate_format(rate); /* try and find a match from the codec and cpu DAI capabilities */ for (j = 0; j < rtd->codec_dai->caps.num_modes; j++) { for (k = 0; k < rtd->cpu_dai->caps.num_modes; k++) { codec_dai_mode = &rtd->codec_dai->caps.mode[j]; cpu_dai_mode = &rtd->cpu_dai->caps.mode[k]; if (!(codec_dai_mode->pcmrate & cpu_dai_mode->pcmrate & pcmrate)) { dbgc("asoc: DAI[%d:%d] failed to match rate\n", j, k); continue; } fmt = codec_dai_mode->fmt & cpu_dai_mode->fmt; if (!(fmt & SND_SOC_DAIFMT_FORMAT_MASK)) { dbgc("asoc: DAI[%d:%d] failed to match format\n", j, k); continue; } if (!(fmt & SND_SOC_DAIFMT_CLOCK_MASK)) { dbgc("asoc: DAI[%d:%d] failed to match clock masters\n", j, k); continue; } if (!(fmt & SND_SOC_DAIFMT_INV_MASK)) { dbgc("asoc: DAI[%d:%d] failed to match invert\n", j, k); continue; } if (!(codec_dai_mode->pcmfmt & cpu_dai_mode->pcmfmt)) { dbgc("asoc: DAI[%d:%d] failed to match pcm format\n", j, k); continue; } if (!(codec_dai_mode->pcmdir & cpu_dai_mode->pcmdir)) { dbgc("asoc: DAI[%d:%d] failed to match direction\n", j, k); continue; } /* todo - still need to add tdm selection */ rtd->cpu_dai->dai_runtime.fmt = rtd->codec_dai->dai_runtime.fmt = 1 << (ffs(fmt & SND_SOC_DAIFMT_FORMAT_MASK) -1) | 1 << (ffs(fmt & SND_SOC_DAIFMT_CLOCK_MASK) - 1) | 1 << (ffs(fmt & SND_SOC_DAIFMT_INV_MASK) - 1); clk_info.bclk_master = rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK; /* make sure the ratio between rate and master * clock is acceptable*/ fs = (cpu_dai_mode->fs & codec_dai_mode->fs); if (fs == 0) { dbgc("asoc: DAI[%d:%d] failed to match FS\n", j, k); continue; } clk_info.fs = rtd->cpu_dai->dai_runtime.fs = rtd->codec_dai->dai_runtime.fs = fs; /* calculate audio system clocking using slowest clocks possible*/ mclk = soc_get_mclk(rtd, &clk_info); if (mclk == 0) { dbgc("asoc: DAI[%d:%d] configuration not clockable\n", j, k); dbgc("asoc: rate %d fs %d master %x\n", rate, fs, clk_info.bclk_master); continue; } /* calculate word size (per channel) and frame size */ rtd->codec_dai->dai_runtime.pcmfmt = rtd->cpu_dai->dai_runtime.pcmfmt = 1 << params_format(params); chn = params_channels(params); /* i2s always has left and right */ if (params_channels(params) == 1 && rtd->cpu_dai->dai_runtime.fmt & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_LEFT_J)) chn <<= 1; /* Calculate bfs - the ratio between bitclock and the sample rate * We must take into consideration the dividers and multipliers * used in the codec and cpu DAI modes. We always choose the * lowest possible clocks to reduce power. */ switch (CODEC_CPU(codec_dai_mode->flags, cpu_dai_mode->flags)) { case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_DIV): /* cpu & codec bfs dividers */ rtd->cpu_dai->dai_runtime.bfs = rtd->codec_dai->dai_runtime.bfs = 1 << (fls(codec_dai_mode->bfs & cpu_dai_mode->bfs) - 1); break; case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_RCW): /* normalise bfs codec divider & cpu rcw mult */ codec_bfs = soc_bfs_div_to_rcw(codec_dai_mode->bfs, rate, mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); rtd->cpu_dai->dai_runtime.bfs = 1 << (ffs(codec_bfs & cpu_dai_mode->bfs) - 1); cpu_bfs = soc_bfs_rcw_to_div(cpu_dai_mode->bfs, rate, mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); rtd->codec_dai->dai_runtime.bfs = 1 << (fls(codec_dai_mode->bfs & cpu_bfs) - 1); break; case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_DIV): /* normalise bfs codec rcw mult & cpu divider */ codec_bfs = soc_bfs_rcw_to_div(codec_dai_mode->bfs, rate, mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); rtd->cpu_dai->dai_runtime.bfs = 1 << (fls(codec_bfs & cpu_dai_mode->bfs) -1); cpu_bfs = soc_bfs_div_to_rcw(cpu_dai_mode->bfs, rate, mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); rtd->codec_dai->dai_runtime.bfs = 1 << (ffs(codec_dai_mode->bfs & cpu_bfs) -1); break; case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_RCW): /* codec & cpu bfs rate rcw multipliers */ rtd->cpu_dai->dai_runtime.bfs = rtd->codec_dai->dai_runtime.bfs = 1 << (ffs(codec_dai_mode->bfs & cpu_dai_mode->bfs) -1); break; case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_RATE): /* normalise cpu bfs rate const multiplier & codec div */ cpu_bfs = soc_bfs_rate_to_div(cpu_dai_mode->bfs, rate, mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); if(codec_dai_mode->bfs & cpu_bfs) { rtd->codec_dai->dai_runtime.bfs = cpu_bfs; rtd->cpu_dai->dai_runtime.bfs = cpu_dai_mode->bfs; } else rtd->cpu_dai->dai_runtime.bfs = 0; break; case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_RATE): /* normalise cpu bfs rate const multiplier & codec rcw mult */ cpu_bfs = soc_bfs_rate_to_rcw(cpu_dai_mode->bfs, rate, mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); if(codec_dai_mode->bfs & cpu_bfs) { rtd->codec_dai->dai_runtime.bfs = cpu_bfs; rtd->cpu_dai->dai_runtime.bfs = cpu_dai_mode->bfs; } else rtd->cpu_dai->dai_runtime.bfs = 0; break; case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_RCW): /* normalise cpu bfs rate rcw multiplier & codec const mult */ codec_bfs = soc_bfs_rate_to_rcw(codec_dai_mode->bfs, rate, mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); if(cpu_dai_mode->bfs & codec_bfs) { rtd->cpu_dai->dai_runtime.bfs = codec_bfs; rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs; } else rtd->cpu_dai->dai_runtime.bfs = 0; break; case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_DIV): /* normalise cpu bfs div & codec const mult */ codec_bfs = soc_bfs_rate_to_div(codec_dai_mode->bfs, rate, mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); if(cpu_dai_mode->bfs & codec_bfs) { rtd->cpu_dai->dai_runtime.bfs = codec_bfs; rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs; } else rtd->cpu_dai->dai_runtime.bfs = 0; break; case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_RATE): /* cpu & codec constant mult */ if(codec_dai_mode->bfs == cpu_dai_mode->bfs) rtd->cpu_dai->dai_runtime.bfs = rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs; else rtd->cpu_dai->dai_runtime.bfs = rtd->codec_dai->dai_runtime.bfs = 0; break; } /* make sure the bit clock speed is acceptable */ if (!rtd->cpu_dai->dai_runtime.bfs || !rtd->codec_dai->dai_runtime.bfs) { dbgc("asoc: DAI[%d:%d] failed to match BFS\n", j, k); dbgc("asoc: cpu_dai %llu codec %llu\n", rtd->cpu_dai->dai_runtime.bfs, rtd->codec_dai->dai_runtime.bfs); dbgc("asoc: mclk %d hwfmt %x\n", mclk, fmt); continue; } goto found; } } printk(KERN_ERR "asoc: no matching DAI found between codec and CPU\n"); return -EINVAL; found: /* we have matching DAI's, so complete the runtime info */ rtd->codec_dai->dai_runtime.pcmrate = rtd->cpu_dai->dai_runtime.pcmrate = soc_get_rate_format(rate); rtd->codec_dai->dai_runtime.priv = codec_dai_mode->priv; rtd->cpu_dai->dai_runtime.priv = cpu_dai_mode->priv; rtd->codec_dai->dai_runtime.flags = codec_dai_mode->flags; rtd->cpu_dai->dai_runtime.flags = cpu_dai_mode->flags; /* for debug atm */ dbg("asoc: DAI[%d:%d] Match OK\n", j, k); if (rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) { codec_bclk = (rtd->codec_dai->dai_runtime.fs * params_rate(params)) / SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); dbg("asoc: codec fs %d mclk %d bfs div %d bclk %d\n", rtd->codec_dai->dai_runtime.fs, mclk, SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk); } else if(rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RATE) { codec_bclk = params_rate(params) * rtd->codec_dai->dai_runtime.bfs; dbg("asoc: codec fs %d mclk %d bfs rate mult %llu bclk %d\n", rtd->codec_dai->dai_runtime.fs, mclk, rtd->codec_dai->dai_runtime.bfs, codec_bclk); } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RCW) { codec_bclk = params_rate(params) * params_channels(params) * snd_pcm_format_physical_width(rtd->codec_dai->dai_runtime.pcmfmt) * SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs); dbg("asoc: codec fs %d mclk %d bfs rcw mult %d bclk %d\n", rtd->codec_dai->dai_runtime.fs, mclk, SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk); } else codec_bclk = 0; if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) { cpu_bclk = (rtd->cpu_dai->dai_runtime.fs * params_rate(params)) / SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); dbg("asoc: cpu fs %d mclk %d bfs div %d bclk %d\n", rtd->cpu_dai->dai_runtime.fs, mclk, SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk); } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RATE) { cpu_bclk = params_rate(params) * rtd->cpu_dai->dai_runtime.bfs; dbg("asoc: cpu fs %d mclk %d bfs rate mult %llu bclk %d\n", rtd->cpu_dai->dai_runtime.fs, mclk, rtd->cpu_dai->dai_runtime.bfs, cpu_bclk); } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RCW) { cpu_bclk = params_rate(params) * params_channels(params) * snd_pcm_format_physical_width(rtd->cpu_dai->dai_runtime.pcmfmt) * SND_SOC_FSBW_REAL(rtd->cpu_dai->dai_runtime.bfs); dbg("asoc: cpu fs %d mclk %d bfs mult rcw %d bclk %d\n", rtd->cpu_dai->dai_runtime.fs, mclk, SND_SOC_FSBW_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk); } else cpu_bclk = 0; /* * Check we have matching bitclocks. If we don't then it means the * sysclock returned by either the codec or cpu DAI (selected by the * machine sysclock function) is wrong compared with the supported DAI * modes for the codec or cpu DAI. Check your codec or CPU DAI * config_sysclock() functions. */ if (cpu_bclk != codec_bclk && cpu_bclk){ printk(KERN_ERR "asoc: codec and cpu bitclocks differ, audio may be wrong speed\n" ); printk(KERN_ERR "asoc: codec %d != cpu %d\n", codec_bclk, cpu_bclk); } switch(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { case SND_SOC_DAIFMT_CBM_CFM: dbg("asoc: DAI codec BCLK master, LRC master\n"); break; case SND_SOC_DAIFMT_CBS_CFM: dbg("asoc: DAI codec BCLK slave, LRC master\n"); break; case SND_SOC_DAIFMT_CBM_CFS: dbg("asoc: DAI codec BCLK master, LRC slave\n"); break; case SND_SOC_DAIFMT_CBS_CFS: dbg("asoc: DAI codec BCLK slave, LRC slave\n"); break; } dbg("asoc: mode %x, invert %x\n", 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\n", runtime->hw.rates); dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); dbg("asoc: min rate %d max rate %d\n", 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(struct work_struct *work) { struct snd_soc_device *socdev = container_of(work, struct snd_soc_device, delayed_work.work); struct snd_soc_codec *codec = socdev->codec; struct snd_soc_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; schedule_delayed_work(&socdev->delayed_work, msecs_to_jiffies(pmdown_time)); } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(codec, 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) { printk(KERN_ERR "asoc: platform prepare error\n"); goto out; } } if (rtd->codec_dai->ops.prepare) { ret = rtd->codec_dai->ops.prepare(substream); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); 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(&socdev->delayed_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_scheduled_work(); 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 */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); return 0; 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 (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, err = 0; mutex_lock(&codec->mutex); for(i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { err = socdev->machine->dai_link[i].init(codec); if (err < 0) { printk(KERN_ERR "asoc: failed to init %s\n", socdev->machine->dai_link[i].stream_name); continue; } } 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); goto out; } #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); 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 entries\n"); out: 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");