October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: wm9713: convert to regmap
by Robert Jarzmik 27 Oct '15

27 Oct '15

Convert the Wolfson WM9713 to regmap API. This will leverage all the regmap functions (debug, registers update, etc ...). As a bonus, this will pave the path to gpio chip introduction, and devicetree support. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: fix suspend/resume (that specific part is not tested yet) --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9713.c | 296 +++++++++++++++++++++++----------------------- 2 files changed, 152 insertions(+), 145 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733ecd17f..ba306f8f3717 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -879,6 +879,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + select REGMAP_AC97 # Amp config SND_SOC_LM4857 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4083a5130cbd..49f2f41dc553 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -19,6 +19,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <linux/device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -39,34 +40,6 @@ struct wm9713_priv { struct mutex lock; }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg); -static int ac97_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val); - -/* - * WM9713 register cache - * Reg 0x3c bit 15 is used by touch driver. - */ -static const u16 wm9713_reg[] = { - 0x6174, 0x8080, 0x8080, 0x8080, - 0xc880, 0xe808, 0xe808, 0x0808, - 0x00da, 0x8000, 0xd600, 0xaaa0, - 0xaaa0, 0xaaa0, 0x0000, 0x0000, - 0x0f0f, 0x0040, 0x0000, 0x7f00, - 0x0405, 0x0410, 0xbb80, 0xbb80, - 0x0000, 0xbb80, 0x0000, 0x4523, - 0x0000, 0x2000, 0x7eff, 0xffff, - 0x0000, 0x0000, 0x0080, 0x0000, - 0x0000, 0x0000, 0xfffe, 0xffff, - 0x0000, 0x0000, 0x0000, 0xfffe, - 0x4000, 0x0000, 0x0000, 0x0000, - 0xb032, 0x3e00, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0006, - 0x0001, 0x0000, 0x574d, 0x4c13, -}; - #define HPL_MIXER 0 #define HPR_MIXER 1 @@ -220,18 +193,15 @@ static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - u16 status, rate; if (WARN_ON(event != SND_SOC_DAPM_PRE_PMD)) return -EINVAL; /* Gracefully shut down the voice interface. */ - status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000; - rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, 0x0200); schedule_timeout_interruptible(msecs_to_jiffies(1)); - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); - ac97_write(codec, AC97_EXTENDED_MID, status); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, 0x0f00); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x1000, 0x1000); return 0; } @@ -674,39 +644,97 @@ static const struct snd_soc_dapm_route wm9713_audio_map[] = { {"Capture Mono Mux", "Right", "Right Capture Source"}, }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm9713_readable_reg(struct device *dev, unsigned int reg) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || - reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || - reg == AC97_CD) - return soc_ac97_ops->read(wm9713->ac97, reg); - else { - reg = reg >> 1; - - if (reg >= (ARRAY_SIZE(wm9713_reg))) - return -EIO; - - return cache[reg]; + switch (reg) { + case AC97_RESET ... AC97_PCM_SURR_DAC_RATE: + case AC97_PCM_LR_ADC_RATE: + case AC97_CENTER_LFE_MASTER: + case AC97_SPDIF ... AC97_LINE1_LEVEL: + case AC97_GPIO_CFG ... 0x5c: + case AC97_CODEC_CLASS_REV ... AC97_PCI_SID: + case 0x74 ... AC97_VENDOR_ID2: + return true; + default: + return false; } } -static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) +static bool wm9713_writeable_reg(struct device *dev, unsigned int reg) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + switch (reg) { + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return false; + default: + return wm9713_readable_reg(dev, reg); + } +} - u16 *cache = codec->reg_cache; - soc_ac97_ops->write(wm9713->ac97, reg, val); - reg = reg >> 1; - if (reg < (ARRAY_SIZE(wm9713_reg))) - cache[reg] = val; +static const struct reg_default wm9713_reg_defaults[] = { + { 0x02, 0x8080 }, /* Speaker Output Volume */ + { 0x04, 0x8080 }, /* Headphone Output Volume */ + { 0x06, 0x8080 }, /* Out3/OUT4 Volume */ + { 0x08, 0xc880 }, /* Mono Volume */ + { 0x0a, 0xe808 }, /* LINEIN Volume */ + { 0x0c, 0xe808 }, /* DAC PGA Volume */ + { 0x0e, 0x0808 }, /* MIC PGA Volume */ + { 0x10, 0x00da }, /* MIC Routing Control */ + { 0x12, 0x8000 }, /* Record PGA Volume */ + { 0x14, 0xd600 }, /* Record Routing */ + { 0x16, 0xaaa0 }, /* PCBEEP Volume */ + { 0x18, 0xaaa0 }, /* VxDAC Volume */ + { 0x1a, 0xaaa0 }, /* AUXDAC Volume */ + { 0x1c, 0x0000 }, /* Output PGA Mux */ + { 0x1e, 0x0000 }, /* DAC 3D control */ + { 0x20, 0x0f0f }, /* DAC Tone Control*/ + { 0x22, 0x0040 }, /* MIC Input Select & Bias */ + { 0x24, 0x0000 }, /* Output Volume Mapping & Jack */ + { 0x26, 0x7f00 }, /* Powerdown Ctrl/Stat*/ + { 0x28, 0x0405 }, /* Extended Audio ID */ + { 0x2a, 0x0410 }, /* Extended Audio Start/Ctrl */ + { 0x2c, 0xbb80 }, /* Audio DACs Sample Rate */ + { 0x2e, 0xbb80 }, /* AUXDAC Sample Rate */ + { 0x32, 0xbb80 }, /* Audio ADCs Sample Rate */ + { 0x36, 0x4523 }, /* PCM codec control */ + { 0x3a, 0x2000 }, /* SPDIF control */ + { 0x3c, 0xfdff }, /* Powerdown 1 */ + { 0x3e, 0xffff }, /* Powerdown 2 */ + { 0x40, 0x0000 }, /* General Purpose */ + { 0x42, 0x0000 }, /* Fast Power-Up Control */ + { 0x44, 0x0080 }, /* MCLK/PLL Control */ + { 0x46, 0x0000 }, /* MCLK/PLL Control */ + { 0x4c, 0xfffe }, /* GPIO Pin Configuration */ + { 0x4e, 0xffff }, /* GPIO Pin Polarity / Type */ + { 0x50, 0x0000 }, /* GPIO Pin Sticky */ + { 0x52, 0x0000 }, /* GPIO Pin Wake-Up */ + /* GPIO Pin Status */ + { 0x56, 0xfffe }, /* GPIO Pin Sharing */ + { 0x58, 0x4000 }, /* GPIO PullUp/PullDown */ + { 0x5a, 0x0000 }, /* Additional Functions 1 */ + { 0x5c, 0x0000 }, /* Additional Functions 2 */ + { 0x60, 0xb032 }, /* ALC Control */ + { 0x62, 0x3e00 }, /* ALC / Noise Gate Control */ + { 0x64, 0x0000 }, /* AUXDAC input control */ + { 0x74, 0x0000 }, /* Digitiser Reg 1 */ + { 0x76, 0x0006 }, /* Digitiser Reg 2 */ + { 0x78, 0x0001 }, /* Digitiser Reg 3 */ + { 0x7a, 0x0000 }, /* Digitiser Read Back */ +}; - return 0; -} +static const struct regmap_config wm9713_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x7e, + .cache_type = REGCACHE_RBTREE, + + .reg_defaults = wm9713_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm9713_reg_defaults), + .volatile_reg = regmap_ac97_default_volatile, + .readable_reg = wm9713_readable_reg, + .writeable_reg = wm9713_writeable_reg, +}; /* PLL divisors */ struct _pll_div { @@ -793,10 +821,8 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, /* turn PLL off ? */ if (freq_in == 0) { /* disable PLL power and select ext source */ - reg = ac97_read(codec, AC97_HANDSET_RATE); - ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0080, 0x0080); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x0200, 0x0200); wm9713->pll_in = 0; return 0; } @@ -806,7 +832,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, if (pll_div.k == 0) { reg = (pll_div.n << 12) | (pll_div.lf << 11) | (pll_div.divsel << 9) | (pll_div.divctl << 8); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); } else { /* write the fractional k to the reg 0x46 pages */ reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) | @@ -814,33 +840,31 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, /* K [21:20] */ reg = reg2 | (0x5 << 4) | (pll_div.k >> 20); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [19:16] */ reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [15:12] */ reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [11:8] */ reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [7:4] */ reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */ - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); } /* turn PLL on and select as source */ - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); - reg = ac97_read(codec, AC97_HANDSET_RATE); - ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x0200, 0x0000); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0080, 0x0000); wm9713->pll_in = freq_in; /* wait 10ms AC97 link frames for the link to stabilise */ @@ -863,10 +887,10 @@ static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai, int tristate) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff; if (tristate) - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x6000, 0x0000); return 0; } @@ -879,36 +903,30 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg; switch (div_id) { case WM9713_PCMCLK_DIV: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, div); break; case WM9713_CLKA_MULT: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0002, div); break; case WM9713_CLKB_MULT: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0004, div); break; case WM9713_HIFI_DIV: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x7000, div); break; case WM9713_PCMBCLK_DIV: - reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff; - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div); + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, 0x0e00, div); break; case WM9713_PCMCLK_PLL_DIV: - reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; - ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div); + snd_soc_update_bits(codec, AC97_LINE1_LEVEL, + 0x007f, div | 0x60); break; case WM9713_HIFI_PLL_DIV: - reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; - ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div); + snd_soc_update_bits(codec, AC97_LINE1_LEVEL, + 0x007f, div | 0x70); break; default: return -EINVAL; @@ -921,7 +939,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5; + u16 gpio = snd_soc_read(codec, AC97_GPIO_CFG) & 0xffc5; u16 reg = 0x8000; /* clock masters */ @@ -974,8 +992,8 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, break; } - ac97_write(codec, AC97_GPIO_CFG, gpio); - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + snd_soc_write(codec, AC97_GPIO_CFG, gpio); + snd_soc_write(codec, AC97_CENTER_LFE_MASTER, reg); return 0; } @@ -984,24 +1002,24 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; + /* enable PCM interface in master mode */ switch (params_width(params)) { case 16: break; case 20: - reg |= 0x0004; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x0004); break; case 24: - reg |= 0x0008; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x0008); break; case 32: - reg |= 0x000c; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x000c); break; } - - /* enable PCM interface in master mode */ - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); return 0; } @@ -1011,17 +1029,15 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; int reg; - u16 vra; - vra = ac97_read(codec, AC97_EXTENDED_STATUS); - ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x0001, 0x0001); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; - return ac97_write(codec, reg, runtime->rate); + return snd_soc_write(codec, reg, runtime->rate); } static int ac97_aux_prepare(struct snd_pcm_substream *substream, @@ -1029,17 +1045,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - u16 vra, xsle; - vra = ac97_read(codec, AC97_EXTENDED_STATUS); - ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); - xsle = ac97_read(codec, AC97_PCI_SID); - ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x0001, 0x0001); + snd_soc_update_bits(codec, AC97_PCI_SID, 0x8000, 0x8000); if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; - return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); + return snd_soc_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); } #define WM9713_RATES (SNDRV_PCM_RATE_8000 | \ @@ -1128,27 +1141,23 @@ static struct snd_soc_dai_driver wm9713_dai[] = { static int wm9713_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: /* enable thermal shutdown */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; - ac97_write(codec, AC97_EXTENDED_MID, reg); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0xe400, 0x0000); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* enable master bias and vmid */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; - ac97_write(codec, AC97_EXTENDED_MID, reg); - ac97_write(codec, AC97_POWERDOWN, 0x0000); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0xc400, 0x0000); + snd_soc_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* disable everything including AC link */ - ac97_write(codec, AC97_EXTENDED_MID, 0xffff); - ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); - ac97_write(codec, AC97_POWERDOWN, 0xffff); + snd_soc_write(codec, AC97_EXTENDED_MID, 0xffff); + snd_soc_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + snd_soc_write(codec, AC97_POWERDOWN, 0xffff); break; } return 0; @@ -1156,16 +1165,17 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec, static int wm9713_soc_suspend(struct snd_soc_codec *codec) { - u16 reg; + regcache_cache_only(codec->component.regmap, true); + snd_soc_cache_sync(codec); /* Disable everything except touchpanel - that will be handled * by the touch driver and left disabled if touch is not in * use. */ - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg | 0x7fff); - ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); - ac97_write(codec, AC97_POWERDOWN, 0x6f00); - ac97_write(codec, AC97_POWERDOWN, 0xffff); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x7fff, + 0x7fff); + snd_soc_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + snd_soc_write(codec, AC97_POWERDOWN, 0x6f00); + snd_soc_write(codec, AC97_POWERDOWN, 0xffff); return 0; } @@ -1173,8 +1183,7 @@ static int wm9713_soc_suspend(struct snd_soc_codec *codec) static int wm9713_soc_resume(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - int i, ret; - u16 *cache = codec->reg_cache; + int ret; ret = snd_ac97_reset(wm9713->ac97, true, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); @@ -1189,12 +1198,8 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) /* only synchronise the codec if warm reset failed */ if (ret == 0) { - for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) { - if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || - i == AC97_EXTENDED_MSTATUS || i > 0x66) - continue; - soc_ac97_ops->write(wm9713->ac97, i, cache[i>>1]); - } + regcache_cache_only(codec->component.regmap, false); + snd_soc_cache_sync(codec); } return ret; @@ -1203,16 +1208,23 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - int reg; + struct regmap *regmap; wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); + regmap = devm_regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); + if (IS_ERR(regmap)) { + snd_soc_free_ac97_codec(wm9713->ac97); + return PTR_ERR(regmap); + } + + snd_soc_codec_init_regmap(codec, regmap); + /* unmute the adc - move to kcontrol */ - reg = ac97_read(codec, AC97_CD) & 0x7fff; - ac97_write(codec, AC97_CD, reg); + snd_soc_update_bits(codec, AC97_CD, 0x7fff, 0x0000); return 0; } @@ -1230,13 +1242,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { .remove = wm9713_soc_remove, .suspend = wm9713_soc_suspend, .resume = wm9713_soc_resume, - .read = ac97_read, - .write = ac97_write, .set_bias_level = wm9713_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm9713_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = wm9713_reg, .controls = wm9713_snd_ac97_controls, .num_controls = ARRAY_SIZE(wm9713_snd_ac97_controls), -- 2.1.4

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[alsa-devel] alsa audio open error
by Srinivasan S 27 Oct '15

27 Oct '15

Dear Alsa community, I facing the below error, am using am335x & ti sdk 8 Something wrong when execute arecord & aplay : arecord and aplay does not work for the first time & it works after throwing error after 5 times Couple of times I get the below error: root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono ^CAborted by signal Interrupt... Aborted by signal Interrupt... root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave arecord: main:722: audio open error: Invalid argument aplay: playback:2715: read error root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dshare.c:710:(snd_pcm_dshare_open) unable to open slave aplay: main:722: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dshare.c:710:(snd_pcm_dshare_open) unable to open slave aplay: main:722: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dshare.c:710:(snd_pcm_dshare_open) unable to open slave aplay: main:722: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dshare.c:710:(snd_pcm_dshare_open) unable to open slave aplay: main:722: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono But there is no answer... Could someone give me some clues? Kindly do the needful as early as possible Many Thanks in advance

1 0

[alsa-devel] No sound after suspend-to-ram
by George Gugulea 27 Oct '15

27 Oct '15

After a fresh reboot, the sound works ok but if I put the computer to sleep (pm-suspend), after it wakes up there is no sound anymore. I did try 'alsa force-reload' after suspend but still no sound. After 'pm-hibernate' sound is back again. I found an old thread here: https://bugzilla.kernel.org/show_bug.cgi?id=49421 where Michael seems to have the same problem which is still unsolved; maybe I can help with more info. I attached the 'alsa-info.sh' output before and after suspend: 'alsa-info.txt.before' and 'alsa-info.txt.after' (as a patch to stay below 80k mail size rule: diff -u alsa-info.txt.before alsa-info.txt.after > alsa-info.txt.patch) The only difference between before and after is this one: @@ -241,7 +241,7 @@ Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x11, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x12 0x12] - Converter: stream=4, channel=0 + Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 The PC is here: http://www.dell.com/us/p/xps-27-2720-aio/pd?ref=PD_OC, but the 2y older version with no touch If I boot into windows (8.1) everything works great, including sound (even after suspend) -- gg

3 15

[alsa-devel] Using snd_pcm_set_params() on a capture handle
by Erik de Castro Lopo 27 Oct '15

27 Oct '15

Hi all, I've got a simple PCM capture test program below, which is basically a slightly tweak version of something from: http://equalarea.com/paul/alsa-audio.html The program uses two different methods of setting up the capture handle; one that works (and is complex) and one that doesn't but seems like it should (and is simple). I'm trying to figure out why the simple one (using snd_pcm_set_params) doesn't work. The program can be compiled and run in two ways: gcc -Wall -O2 -DUSE_SET_PARAMS=1 alsa_capture.c -lasound -o alsa_capture ./alsa_capture which fails with: Read from audio interface failed (Input/output error) and: gcc -Wall -O2 -DUSE_SET_PARAMS=0 alsa_capture.c -lasound -o alsa_capture ./alsa_capture which works as expected. Questions: * Is snd_pcm_set_params() supposed to work on capture handles (I've used it successfully on playback handles)? * Is my use of snd_pcm_set_params() missing something and if so what? * If snd_pcm_set_params() is not supposed to work on capture handles would it be possible to update the documentation to reflect that? Thanks, Erik //--------------------------------------------- #include <stdio.h> #include <stdlib.h> #include <alsa/asoundlib.h> #include <alsa/pcm.h> int main (int argc, char *argv[]) { int i; int err; char *buffer; int buffer_frames = 128; unsigned int rate = 44100; snd_pcm_t *capture_handle; snd_pcm_format_t format = SND_PCM_FORMAT_S32_LE; char* device = "default"; if (argc > 1) device = argv[1]; if ((err = snd_pcm_open (&capture_handle, device, SND_PCM_STREAM_CAPTURE, 0)) < 0) { printf ("Cannot open audio device %s (%s)\n", device, snd_strerror (err)); exit (1) ; } if (USE_SET_PARAMS) { if ((err = snd_pcm_set_params (capture_handle, format, SND_PCM_ACCESS_RW_INTERLEAVED, 2, rate, 0, 500000)) < 0) { /* 0.5sec */ printf ("Capture open error: %s\n", snd_strerror (err)); exit (1) ; } } else { snd_pcm_hw_params_t *hw_params; if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { printf ("Cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_any (capture_handle, hw_params)) < 0) { printf ("Cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_set_access (capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { printf ("cannot set access type (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_set_format (capture_handle, hw_params, format)) < 0) { printf ("Cannot set sample format (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, hw_params, &rate, 0)) < 0) { printf ("Cannot set sample rate (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, 2)) < 0) { printf ("cannot set channel count (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params (capture_handle, hw_params)) < 0) { printf ("cannot set parameters (%s)\n", snd_strerror (err)); exit (1) ; } snd_pcm_hw_params_free (hw_params); } buffer = malloc (buffer_frames * snd_pcm_format_width(format) / 8 * 2); if ((err = snd_pcm_prepare (capture_handle)) < 0) { printf ("cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1) ; } for (i = 0; i < 10; ++i) { if ((err = snd_pcm_readi (capture_handle, buffer, buffer_frames)) != buffer_frames) { printf ("Read from audio interface failed (%s)\n", snd_strerror (err)); exit (1) ; break; } printf("read %d done\n", i); } free(buffer); snd_pcm_close (capture_handle); puts ("Audio interface closed"); return 0; } -- ---------------------------------------------------------------------- Erik de Castro Lopo http://www.mega-nerd.com/

4 10

[alsa-devel] [PATCH 0/18] ASoC: rsnd: use mod base common method on DMA/CMD/SSIU/SSI-parent
by Kuninori Morimoto 27 Oct '15

27 Oct '15

Hi Mark Renesas sound has (SSI/SSIU/SRC/CTU/MIX/DVC/CMD/AudioDMAC/AudioDMACpp). Here, SSI/SRC/CTU/MIX/DVC are implemented as "module". But, SSI parent / SSIU are implemented as part of SSI, CMD is implemented as part of CTU/MIX/DVC AudioDMAC/AudioDMACpp are implemented as part of SSI/SRC. because I assumed that SSI/SRC/CTU/MIX/DVC (which are implemented as module) are MAIN modules, and others are option. But, current style is 1) ununderstandable each IPs setting order when debugging, because debug message can't indicate invisible "modules". 2) there is no flexibility for each devices. This patch-set makes DMA/CMD/SSIU/SSI-parent as common module. All devices are controlled as common module now. 1) - 12) prepare for each device 13) - 18) mod base common method patches. Becase DMA was implemented as part of SSI/SRC, it was difficult to separate it in 1 patch. It used 3 phases here Kuninori Morimoto (18): 1) ASoC: rsnd: DMA become SSI/SRC member 2) ASoC: rsnd: DMA related definition goes to dma.c 3) ASoC: rsnd: rename rsnd_dma_init() to rsnd_dma_attach() 4) ASoC: rsnd: enable to use rsnd_dai_connect() from each mod 5) ASoC: rsnd: remove all modules when DMA fail case 6) ASoC: rsnd: fixup rsnd_dma_of_path method for mod base common method 7) ASoC: rsnd: move rsnd_src_ssi_irq_enable/disable() to ssi.c 8) ASoC: rsnd: disable SRC.out only when stop timing 9) ASoC: rsnd: rsnd_dai_stream has each mod's status insted of rsnd_mod 10) ASoC: rsnd: Don't stop HW even if a large number of errors occur 11) ASoC: rsnd: avoid pointless loop in rsnd_mod_interrupt() 12) ASoC: rsnd: use common rsnd_ssi_status_xxx() 13) ASoC: rsnd: use mod base common method on DMA phase1 14) ASoC: rsnd: use mod base common method on DMA phase2 15) ASoC: rsnd: use mod base common method on DMA phase3 16) ASoC: rsnd: use mod base common method on CMD 17) ASoC: rsnd: use mod base common method on SSIU 18) ASoC: rsnd: use mod base common method on SSI-parent sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/cmd.c | 153 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/core.c | 155 +++++++++++++------------------------------------ sound/soc/sh/rcar/ctu.c | 8 +++ sound/soc/sh/rcar/dma.c | 240 +++++++++++++++++++++++++++++++++++++++++++++------------------------------ sound/soc/sh/rcar/dvc.c | 18 ++++-- sound/soc/sh/rcar/mix.c | 10 +++- sound/soc/sh/rcar/rsnd.h | 88 +++++++++++++++------------- sound/soc/sh/rcar/src.c | 179 ++++++++++++-------------------------------------------- sound/soc/sh/rcar/ssi.c | 423 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++-------------------------------------------------------------- sound/soc/sh/rcar/ssiu.c | 181 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 11 files changed, 860 insertions(+), 597 deletions(-)

2 36

[alsa-devel] [PATCH 0/5] ALSA: firewire-tascam: add MIDI functionality
by Takashi Sakamoto 27 Oct '15

27 Oct '15

Hi, TASCAM FireWire series driver may be newly available in Linux 4.4. For the driver, this patchset adds ALSA MIDI ports to communicate to physical MIDI ports. This patchset includes some fixes, and updates my former post: [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html Changes: * support system exclusive messages * drop virtual MIDI ports * drop MMAP support via hwdep interface for status and control message Although TASCAM FireWire series has physical controls, this patchset doesn't support them. I think some facilities are required to enable this functionality: * handling MIDI messages on virtual MIDI ports by this driver * handling status and control messages in isochronous packet by this driver * parsing the status and control messages by userspace application * supporting assignment of physical controls to MIDI ports by userspace application * turn on/off LEDs on device by userspace application These work are apparently beyond my current effort. Takashi Sakamoto (5): ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction ALSA: firewire-tascam: add support for outgoing MIDI messages by asynchronous transaction ALSA: firewire-tascam: add support for MIDI functionality ALSA: firewire-tascam: Turn on/off FireWire LED ALSA: firewire-tascam: change device probing processing sound/firewire/tascam/Makefile | 3 +- sound/firewire/tascam/tascam-midi.c | 135 +++++++++++++ sound/firewire/tascam/tascam-transaction.c | 293 +++++++++++++++++++++++++++++ sound/firewire/tascam/tascam.c | 89 +++++---- sound/firewire/tascam/tascam.h | 30 +++ 5 files changed, 503 insertions(+), 47 deletions(-) create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-transaction.c -- 2.1.4

4 25

[alsa-devel] [PATCH v2 0/4] Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 26 Oct '15

26 Oct '15

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver as present on the A10, A20 and A31 SoC boards. For now just the SPDIF transmitter has been tested on a Mele A2000. In order for this patch set to be functional we require audio clock patches which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery This is version 2 of the patch series. After Mark's initial comments I realised that I shouldn't of blindly copied the register set-up from the original SDK that was released. It doesn't help also that the datasheet that finally emerged lacked a serious amount of detail. If anybody does recognise the IP used for the block then please speak out. In short, much of the code that was in the first release is no longer here. I'm still unsure about the machine driver setup. Simple card seems a bit overkill and what I've added is pretty much a copy of what is done in imx-spdif. I'm not planning on arguing the case for my method so if you can advise then I'm all ears. Thanks in advance, CK --- Changes since v1: - Moved sunxi-machine-spdif.c to seperate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into seperate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. --- Marcus Cooper (4): dt-bindings: add sunxi SPDIF transceiver bindings dt-binding: Add sunxi S/PDIF machine driver ASoC: sunxi: Add S/PDIF machine driver. ASOC: sunxi: Add support for the spdif block .../devicetree/bindings/sound/sunxi,spdif.txt | 49 ++ .../bindings/sound/sunxi-audio-spdif.txt | 36 ++ sound/soc/sunxi/Kconfig | 12 + sound/soc/sunxi/Makefile | 4 + sound/soc/sunxi/sun4i-spdif.c | 612 +++++++++++++++++++++ sound/soc/sunxi/sunxi-machine-spdif.c | 108 ++++ 6 files changed, 821 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/sunxi-audio-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c create mode 100644 sound/soc/sunxi/sunxi-machine-spdif.c -- 1.9.1

4 17

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Convert leftover pr_info() and pr_err()
by Takashi Iwai 26 Oct '15

26 Oct '15

Use codec_*() macro instead of pr_*() for improving the log information. The current form even slips from alsa-info.sh. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 3a02e5c14d09..f8a12ca477f1 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2673,13 +2673,13 @@ static bool dspload_wait_loaded(struct hda_codec *codec) do { if (dspload_is_loaded(codec)) { - pr_info("ca0132 DOWNLOAD OK :-) DSP IS RUNNING.\n"); + codec_info(codec, "ca0132 DSP downloaded and running\n"); return true; } msleep(20); } while (time_before(jiffies, timeout)); - pr_err("ca0132 DOWNLOAD FAILED!!! DSP IS NOT RUNNING.\n"); + codec_err(codec, "ca0132 failed to download DSP\n"); return false; } @@ -4375,7 +4375,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { - pr_err("ca0132 dspload_image failed.\n"); + codec_err(codec, "ca0132 DSP load image failed\n"); goto exit_download; } -- 2.6.2

1 0

Re: [alsa-devel] BAT: restructuring code
by Caleb Crome 26 Oct '15

26 Oct '15

On Fri, Oct 23, 2015 at 3:47 AM, Liam Girdwood <liam.r.girdwood(a)linux.intel.com> wrote: > Hi Caleb, > > I have no problem as long as we can still support Android (via tinyalsa) > and Chrome (via alsa-lib) and also tinycompress. You will need to > justify any changes you make in your commit message. > > Btw, probably best to also CC alsa-devel mailing list for these sort of > questions too as other folks doing BAT patches may also be interested. > > Fwiw, a structure like your proposing could also be used for SIMD based > optimisations too (if we ever need them). > > Liam > > On Thu, 2015-10-22 at 17:23 -0700, Caleb Crome wrote: >> Hi guys, >> since i've got several different tests to write, I'm basically >> rewriting a meaningful section of the code to make the >> init/generate/playout/analyze modular. >> >> To that end, I'm creating a struct like this: >> >> >> struct bat_ops { >> (void *)(*init)(int channels, int format, float sample_rate, float >> *frequencies) >> // generate_playout_data is called every period to, well, generate >> the data to be played. >> int (*generate_playout_data)(void *priv, int frames, void *buffer); >> // process_capture_data is called every period to do something >> with the data. >> int (*process_input_data)(void *priv, int frames, const void *buffer); >> // analyze is called at the end of the whole capture time. Can be >> used to analyze and report results. >> int (*analyze)(void *priv); >> void (*free)(**priv); >> }; >> >> Then, we can create a separate ops struct for each type of test, i.e. >> a generated sine wave, reading from a file, ramp test, etc. >> >> The alsa code is fantastically complex, so I'm not touching any of >> that. :-/ Just splitting out the generate, capture, and analysis to >> be modular. >> >> Let me know if you think this is going to be a problem. >> >> -Caleb > > In order to support the molecularity, I need the ability for each module (i.e. test type) to accept its own arguments. In order to accomplish this, I think it would be better to use argp instead of getopt_long. Is it okay if I switch it over? Will that break android? Basically, I need the general options like sample rate, channels and format to be in the first set of arguments, but then be able to call a second parser to manage the module's options, like sine frequency (for the sine based test), or impulse response length (for an impulse response test). Perhaps there's a way to do it with getopt_long, but it doesn't look too simple because getopt_long requires all the options to be already defined before parsing starts, whereas argp can bring in new parsers (from the ops struct previously described). Any thoughts? -Caleb

2 2

[alsa-devel] [GIT PULL] ASoC updates for v4.3
by Mark Brown 26 Oct '15

26 Oct '15

The following changes since commit 32b88194f71d6ae7768a29f87fbba454728273ee: Linux 4.3-rc7 (2015-10-25 10:39:47 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.3-rc6 for you to fetch changes up to bb9a13a0209c56cdf27d125a1f2f6f34378c64f4: Merge remote-tracking branches 'asoc/topic/wm8904', 'asoc/topic/wm8955' and 'asoc/topic/wm8960' into asoc-next (2015-10-26 11:16:20 +0900) ---------------------------------------------------------------- ASoC: Updates for v4.4 Not much core work here, a few small tweaks to interfaces but mainly the changes here are driver ones. Highlights include: - Updates to the topology userspace interface - Big updates to the Renesas support from Morimoto-san - Most of the support for Intel Sky Lake systems. - New drivers for Asahi Kasei Microdevices AK4613, Allwinnner A10, Cirrus Logic WM8998, Dialog DA7219, Nuvoton NAU8825 and Rockchip S/PDIF. ---------------------------------------------------------------- Also a new driver for the Atmel Class D speaker drivers which I forgot in the tag message, sorry. Adam Thomson (11): ASoC: Add SOC_DOUBLE_R_EXT ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec ASoC: da7219: Add entry to cover DA7219 bindings document ASoC: da7219: Improve error handling for regulator supplies ASoC: da7219: Use of_match_ptr() when assigning match table ASoC: da7219: Improve error checking of mclk enable/disable ASoC: da7213: Add DT support to codec driver ASoC: da7213: Add support to handle mclk data provided to driver ASoC: da7213: Add bindings documentation for codec driver ASoC: da7213: Correct units description of dmic-clkrate Anatol Pomozov (2): ASoC: nau8825: Add driver for headset chip Nuvoton 8825 ASoC: Document DAI signal polarity Andy Shevchenko (1): ASoC: Intel: use dw_dmac autoconfiguration Axel Lin (20): ASoC: mediatek: Convert to devm_snd_soc_register_card ASoC: au1x: Convert to devm_snd_soc_register_card ASoC: blackfin: Convert to devm_snd_soc_register_card ASoC: mxs-sgtl5000: Convert to devm_snd_soc_register_card ASoC: pxa: Convert to devm_snd_soc_register_card ASoC: sh: siu_dai: Convert to use resource managed APIs ASoC: intel: broadwell: Convert to devm_snd_soc_register_card ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: rockchip: Fix missing CLKDEV_LOOKUP dependency ASoC: rl6347a: Clean up unneeded inclusion of header files ASoC: ad193x-spi: Fixup ad193x_spi_id table ASoC: rt286: Fix run time error while modifying const data ASoC: rt298: Make rt298_index_def const ASoC: rockchip: spdif: Convert to use devm_snd_dmaengine_pcm_register ASoC: ad193x-spi: Add adau1328 to ad193x_spi_id table ASoC: ad193x: Drop .volatile_reg implementation ASoC: ssm2518: Drop .volatile_reg implementation ASoC: da7219: Fix da7219->alc_en state when enabling ALC ASoC: Intel: Skylake: Convert to devm_snd_soc_register_card ASoC: rt5645: Sort the order for register bit defines Bard Liao (1): ASoC: rt298: set register non-volatile by default Ben Zhang (3): ASoC: rt5645: Allow 4 channel recording on AIF1 ASoC: nau8825: Add FLL configuration ASoC: nau8825: Show device properties Charles Keepax (4): ASoC: arizona: Add default cases for event switches ASoC: arizona: Add utility function to check if an input is analog ASoC: Add SOC_SINGLE_RANGE_EXT_TLV macro ASoC: wm5110: Add additional analogue input enable for early revs Cyrille Pitchen (1): ASoC: ad193x: add support to ad1934 Dan Carpenter (7): ASoC: wm8904: harmless underflow in wm8904_put_deemph() ASoC: wm8731: harmless underflow in wm8731_put_deemph() ASoC: wm8960: harmless underflow in wm8960_put_deemph() ASoC: wm8903: harmless underflow in wm8903_put_deemph() ASoC: wm8955: harmless underflow in wm8955_put_deemph() ASoC: es8328: harmless underflow in es8328_put_deemph() ASoC: wm2000: a couple harmless underflows Dan Williams (1): ASoC: Intel: switch from ioremap_cache to memremap Daniel Mack (1): ASoC: pxa: pxa-pcm-lib: switch over to snd-soc-dmaengine-pcm Emilio López (1): ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs Fang, Yang A (1): ASoC: nau8825: add acpi match ID Geert Uytterhoeven (1): ASoC: rsnd: Remove obsolete platform data support Jarkko Nikula (1): spi: atmel: Remove needless bits_per_word and speed_hz tests Jeeja KP (19): ASoC: Intel: Skylake: Remove unused CPU dais ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Check CPA bit in DSP core power down ASoC: Intel: Skylake: Verify the status bit before handling interrupt ASoC: Intel: Skylake: Correct the runtime handler behaviour ASoC: Intel: Skylake: power down all link in suspend ASoC: dapm: Add kcontrol support for PGAs ASoC: dapm: Add startup & shutdown for dai_links ASoC: Intel: Skylake: Fix return for pm_runtime_get_sync() ASoC: Intel: Skylake: Remove BE copier blob memcpy ASoC: Intel: Skylake: Update for ssp node index in copier cfg ASoC: Intel: Skylake: Add support to disable module notifications ASoC: Intel: Skylake: Fix to check return value of dsp init ASoC: Intel: Skylake: Fix to set Non gateway copier cfg ASoC: Intel: Skylake: Add support to topology for module static pin Jie Yang (2): ASoC: compress: add config item for soc-compress to make it compiled only when needed ASoC: Intel: sst: only use sst-firmware when DW DMAC is available John Lin (3): ASoC: rt5645: Add dmi for Broadwell ASoC: rt5645: Add jd_invert for Broadwell ASoC: rt5645: Add dmi "Google Reks" for chrome Julia Lawall (1): ASoC: qcom: change PTR_ERR argument Jyri Sarha (6): ASoC: tlv320aic3x: Improve tdm support ASoC: davinci-mcasp: Add set_tdm_slots() support ASoC: simple-card: Add tdm slot mask support to simple-card ASoC: soc-core: Fix sparse warning in be32_to_cpup() call ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data ASoC: hdmi: Remove obsolete dummy HDMI codec Kuninori Morimoto (31): ASoC: ak4642: use *dev on ak4642_i2c_probe() ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT ASoC: rsnd: tidyup rsnd_dma_to_xxx() position ASoC: rsnd: rename rsnd_src_pcm_new() to rsnd_src_pcm_new_gen2() ASoC: rsnd: remove unneeded sh_clk header ASoC: rsnd: add missing #sound-dai-cells explain on Document ASoC: rsnd: Add Gen3 initial support ASoC: rsnd: add rsnd_mod_get() macro and use it ASoC: rsnd: ADG uses mod base common method ASoC: rsnd: add common mod confirm method ASoC: rsnd: ADG initial setup on rsnd_adg_ssi_clk_init() ASoC: rsnd: tidyup ADG clock calculate method ASoC: rsnd: tidyup ADG debug message for clock selection ASoC: rsnd: adg: ignore undefined clock error ASoC: rsnd: add AUDIO_CLKOUT support ASoC: rsnd: SND_SOC_RCAR depends on COMMON_CLK ASoC: ak4642: exchange macro name to avoid redefinition ASoC: rsnd: fixup clkout_name[] index error for single AUDIO_CLKOUT SoC ASoC: add ak4613 support ASoC: ak4613: add Digital Playback Volume support ASoC: rsnd: Announce for removing Gen1 SRU support ASoC: rsnd: Gen1 probe is not error ASoC: rsnd: fixup print debug message after read ASoC: rsnd: fixup struct rsnd_gen::res array size ASoC: rsnd: fixup devm_request_irq() option on ssi.c ASoC: rsnd: fixup rsnd_mod_call() behavior for debug ASoC: rsnd: fixup rsnd_dai_call() behavior for unimplemented method ASoC: rsnd: fixup rsnd_dai_call() behavior for .stop/.quit ASoC: rsnd: rename rsnd_mod_hw_start/stop to rsnd_mod_power_on/off ASoC: rsnd: remove unused rsnd_dma_to_ssi() macro ASoC: rsnd: remove duplicate parameter from rsnd_ssi_xxx() Lars-Peter Clausen (4): ASoC: Let snd_soc_limit_volume() take a snd_soc_card ASoC: h1940_uda1380: Constify rate constraints ASoC: rx1950_uda1380: Constify rate constraints ASoC: fsl_sai: Constify rate constraints Luis de Bethencourt (8): ASoC: atmel_wm8904: Fix module autoload for OF platform driver ASoC: fsl-asoc-card: Fix module autoload for OF platform driver ASoC: fsl_sai: Fix module autoload for OF platform driver ASoC: jz4740: Fix module autoload for OF platform driver ASoC: kirkwood: Fix module autoload for OF platform driver ASoC: pxa: Fix module autoload for OF platform drivers ASoC: mop500: Fix module autoload for OF platform driver ASoC: ux500: Fix module autoload for OF platform driver Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: add AC'97 support Mark Brown (27): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4613', 'asoc/topic/ak4642', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/hdmi' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/jz4740', 'asoc/topic/kirkwood', 'asoc/topic/mtk', 'asoc/topic/multi' and 'asoc/topic/mxs' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar' and 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/spear', 'asoc/topic/sunxi' and 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge tag 'asoc-fix-v4.3-rc2' into asoc-pxa Merge branch 'fix/rt298' of git://git.kernel.org/.../broonie/sound into asoc-rt298 Merge branch 'topic/fsl-mega-fast' of git://git.kernel.org/.../broonie/sound into asoc-fsl Merge tag 'asoc-v4.3-rc2' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/doc' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ad193x' and 'asoc/topic/arizona' into asoc-next Merge remote-tracking branches 'asoc/topic/atmel-classd' and 'asoc/topic/da7213' into asoc-next Merge remote-tracking branch 'asoc/topic/da7219' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/es8328', 'asoc/topic/fsl' and 'asoc/topic/fsl-card' into asoc-next Merge remote-tracking branches 'asoc/topic/imx' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/nau8825' and 'asoc/topic/pxa' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/rl6347a', 'asoc/topic/rockchip' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt298', 'asoc/topic/rt5640', 'asoc/topic/samsung' and 'asoc/topic/sh' into asoc-next Merge remote-tracking branches 'asoc/topic/ssm2518' and 'asoc/topic/sunxi' into asoc-next Merge remote-tracking branches 'asoc/topic/topology', 'asoc/topic/wm2000', 'asoc/topic/wm8731' and 'asoc/topic/wm8903' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8904', 'asoc/topic/wm8955' and 'asoc/topic/wm8960' into asoc-next Maruthi Srinivas Bayyavarapu (2): ASoC: dwc: support dw i2s in slave mode ASoC: dwc: add check for master/slave format Masanari Iida (1): ASoC: sh: Fit typo in Kconfig Maxime Ripard (4): ASoC: sunxi: Add the Allwinner A10 codec bindings ASoC: sun4i-codec: pass through clk_set_rate error ASoC: sun4i-codec: Remove redundant SND_PCM_RATE_KNOT ASoC: sun4i-codec: Remove the routing property Mengdong Lin (2): ASoC: topology: ABI - Change stream formats to a bitwise flag ASoC: topology: ABI - Remove tdm_slot & dai_fmt from snd_soc_tplg_stream Mythri P K (1): ASoC: dapm: Add snd_soc_dapm_kcontrol_widget() Nicolin Chen (1): ASoC: fsl: Use #ifdef instead of #if for CONFIG_PM_SLEEP Oder Chiou (7): ASoC: rt5645: Remove the repeated definitions ASoC: rt5645: Add the control of ClassD modulator Speaker Gain Ratio ASoC: rt5645: Prevent the weird sound of the headphone while rebooting ASoC: rt5645: Recheck the jack detect status after resuming from S3 ASoC: rt5640: Revise the input pin name of IN1 and IN2 in document of the devicetree ASoC: rt5640: Fill up the IN3's support ASoC: rt5645: Add the HWEQ for the speaker output Omair M Abdullah (1): ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver Ricard Wanderlof (1): ASoC: Handle multiple codecs with split playback / capture Richard Fitzgerald (2): ASoC: arizona: add defines for single-input gain control ASoC: wm8998: Initial WM8998 codec driver Sjoerd Simons (4): ASoC: rockchip: namespace rockchip i2s module name ASoC: dt-bindings: add rockchip tranceiver bindings ASoC: rockchip: Add rockchip SPDIF transceiver driver ASoC: rockchip: Drop unneeded properties rockchip i2s/spdif bindings Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier ASoC: atmel-classd: DT binding for Class D audio amplifier driver Subhransu S. Prusty (2): ASoC: Intel: Skylake: Fix to use correct macros for the path iteration ASoC: Intel: Skylake: Fix to fill all sink/source pipe params Sudip Mukherjee (2): ASoC: Intel: remove unused function ASoC: rt5645: fix build warning Sugar Zhang (3): ASoC: rockchip: i2s: add 8 channels capture support ASoC: rockchip: add capture property ASoC: rockchip: i2s: share tx/rx lrck when symmetric_rates enabled Vedang Patel (6): ASoC: topology: ABI - Add PCM Support and bump ABI version to 4 ASoC: topology: ABI - Remove unused struct snd_soc_tplg_stream_config ASoC: topology: ABI - Use __le32 instead of __u32 in snd_soc_tplg_dapm_widget ASoC: topology: ABI - Add name element to snd_soc_tplg_stream ASoC: topology: ABI - Add the type for BE DAI link ASoC: topology: ABI - Add configuration for BE & Codec-Codec DAI Links Vinod Koul (4): ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Modify the log level Zidan Wang (5): ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast ASoC: fsl_spdif: Add driver suspend and resume to support MEGA Fast ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast ASoC: fsl_esai: Add driver suspend and resume to support MEGA Fast kbuild test robot (2): ASoC: sunxi: fix platform_no_drv_owner.cocci warnings ASoC: nau8825: fix platform_no_drv_owner.cocci warnings Documentation/devicetree/bindings/sound/ak4613.txt | 17 + Documentation/devicetree/bindings/sound/ak4642.txt | 22 +- .../devicetree/bindings/sound/atmel-classd.txt | 52 + Documentation/devicetree/bindings/sound/da7213.txt | 41 + Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- .../devicetree/bindings/sound/nau8825.txt | 102 + .../devicetree/bindings/sound/renesas,rsnd.txt | 7 + .../devicetree/bindings/sound/rockchip-i2s.txt | 6 +- .../devicetree/bindings/sound/rockchip-spdif.txt | 40 + Documentation/devicetree/bindings/sound/rt5640.txt | 9 +- .../devicetree/bindings/sound/sun4i-codec.txt | 27 + .../devicetree/bindings/sound/tdm-slot.txt | 11 +- MAINTAINERS | 1 + drivers/spi/spi-atmel.c | 21 +- include/sound/da7213.h | 3 - include/sound/da7219-aad.h | 99 + include/sound/da7219.h | 55 + include/sound/designware_i2s.h | 2 + include/sound/pxa2xx-lib.h | 1 - include/sound/rt5640.h | 3 +- include/sound/rt5645.h | 2 + include/sound/simple_card.h | 2 + include/sound/soc-dai.h | 19 +- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 27 +- include/uapi/sound/asoc.h | 76 +- sound/arm/pxa2xx-ac97.c | 13 +- sound/arm/pxa2xx-pcm-lib.c | 201 +- sound/arm/pxa2xx-pcm.c | 12 +- sound/arm/pxa2xx-pcm.h | 2 - sound/soc/Kconfig | 6 +- sound/soc/Makefile | 4 +- sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 679 +++++++ sound/soc/atmel/atmel-classd.h | 120 ++ sound/soc/atmel/atmel_wm8904.c | 1 + sound/soc/au1x/db1000.c | 10 +- sound/soc/au1x/db1200.c | 10 +- sound/soc/blackfin/bf5xx-ad1836.c | 11 +- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +- sound/soc/codecs/Kconfig | 23 +- sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ad193x-i2c.c | 8 +- sound/soc/codecs/ad193x-spi.c | 16 +- sound/soc/codecs/ad193x.c | 128 +- sound/soc/codecs/ad193x.h | 9 +- sound/soc/codecs/ak4613.c | 497 +++++ sound/soc/codecs/ak4642.c | 153 +- sound/soc/codecs/arizona.c | 49 + sound/soc/codecs/arizona.h | 14 +- sound/soc/codecs/da7213.c | 190 +- sound/soc/codecs/da7213.h | 8 +- sound/soc/codecs/da7219-aad.c | 823 ++++++++ sound/soc/codecs/da7219-aad.h | 212 +++ sound/soc/codecs/da7219.c | 1955 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 ++++++++ sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdmi.c | 109 -- sound/soc/codecs/nau8825.c | 1309 +++++++++++++ sound/soc/codecs/nau8825.h | 341 ++++ sound/soc/codecs/rl6347a.c | 19 +- sound/soc/codecs/rl6347a.h | 2 + sound/soc/codecs/rt286.c | 9 +- sound/soc/codecs/rt298.c | 11 +- sound/soc/codecs/rt5640.c | 22 +- sound/soc/codecs/rt5645.c | 172 +- sound/soc/codecs/rt5645.h | 30 +- sound/soc/codecs/ssm2518.c | 6 - sound/soc/codecs/tlv320aic3x.c | 30 +- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm5110.c | 187 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8998.c | 1430 ++++++++++++++ sound/soc/codecs/wm8998.h | 23 + sound/soc/davinci/davinci-mcasp.c | 305 +-- sound/soc/dwc/designware_i2s.c | 123 +- sound/soc/fsl/fsl-asoc-card.c | 141 +- sound/soc/fsl/fsl_esai.c | 88 + sound/soc/fsl/fsl_sai.c | 38 +- sound/soc/fsl/fsl_spdif.c | 73 + sound/soc/fsl/fsl_ssi.c | 102 + sound/soc/fsl/imx-spdif.c | 1 + sound/soc/generic/simple-card.c | 8 +- sound/soc/intel/Kconfig | 15 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 19 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/broadwell.c | 9 +- sound/soc/intel/boards/skl_rt286.c | 259 +++ sound/soc/intel/common/Makefile | 6 +- sound/soc/intel/common/sst-dsp-priv.h | 1 + sound/soc/intel/common/sst-dsp.c | 2 + sound/soc/intel/common/sst-dsp.h | 2 + sound/soc/intel/common/sst-firmware.c | 10 +- sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-messages.c | 44 +- sound/soc/intel/skylake/skl-nhlt.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 176 +- sound/soc/intel/skylake/skl-sst-dsp.c | 7 +- sound/soc/intel/skylake/skl-sst-ipc.c | 12 + sound/soc/intel/skylake/skl-sst-ipc.h | 1 + sound/soc/intel/skylake/skl-sst.c | 56 +- sound/soc/intel/skylake/skl-topology.c | 1252 +++++++++++++ sound/soc/intel/skylake/skl-topology.h | 36 +- sound/soc/intel/skylake/skl-tplg-interface.h | 83 + sound/soc/intel/skylake/skl.c | 32 +- sound/soc/intel/skylake/skl.h | 17 +- sound/soc/jz4740/jz4740-i2s.c | 1 + sound/soc/kirkwood/armada-370-db.c | 1 + sound/soc/mediatek/mt8173-max98090.c | 11 +- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 11 +- sound/soc/mxs/mxs-sgtl5000.c | 6 +- sound/soc/omap/rx51.c | 2 +- sound/soc/pxa/brownstone.c | 9 +- sound/soc/pxa/corgi.c | 11 +- sound/soc/pxa/e740_wm9705.c | 5 +- sound/soc/pxa/e750_wm9705.c | 5 +- sound/soc/pxa/e800_wm9712.c | 5 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 11 +- sound/soc/pxa/mioa701_wm9713.c | 11 +- sound/soc/pxa/palm27x.c | 9 +- sound/soc/pxa/poodle.c | 11 +- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 49 +- sound/soc/pxa/pxa2xx-i2s.c | 3 + sound/soc/pxa/pxa2xx-pcm.c | 33 +- sound/soc/pxa/spitz.c | 5 +- sound/soc/pxa/tosa.c | 5 +- sound/soc/pxa/ttc-dkb.c | 12 +- sound/soc/qcom/lpass-cpu.c | 3 +- sound/soc/rockchip/Kconfig | 12 +- sound/soc/rockchip/Makefile | 6 +- sound/soc/rockchip/rockchip_i2s.c | 48 +- sound/soc/rockchip/rockchip_i2s.h | 16 + sound/soc/rockchip/rockchip_spdif.c | 405 ++++ sound/soc/rockchip/rockchip_spdif.h | 63 + sound/soc/samsung/h1940_uda1380.c | 5 +- sound/soc/samsung/rx1950_uda1380.c | 5 +- sound/soc/sh/Kconfig | 3 +- sound/soc/sh/rcar/adg.c | 306 ++- sound/soc/sh/rcar/core.c | 66 +- sound/soc/sh/rcar/ctu.c | 16 +- sound/soc/sh/rcar/dma.c | 2 +- sound/soc/sh/rcar/dvc.c | 16 +- sound/soc/sh/rcar/gen.c | 14 +- sound/soc/sh/rcar/mix.c | 16 +- {include/sound => sound/soc/sh/rcar}/rcar_snd.h | 1 - sound/soc/sh/rcar/rsnd.h | 29 +- sound/soc/sh/rcar/src.c | 27 +- sound/soc/sh/rcar/ssi.c | 113 +- sound/soc/sh/siu_dai.c | 85 +- sound/soc/soc-compress.c | 12 +- sound/soc/soc-core.c | 29 +- sound/soc/soc-dapm.c | 59 +- sound/soc/soc-ops.c | 8 +- sound/soc/soc-pcm.c | 49 + sound/soc/soc-topology.c | 4 +- sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-codec.c | 712 +++++++ sound/soc/ux500/mop500.c | 1 + sound/soc/ux500/ux500_msp_dai.c | 1 + 170 files changed, 14388 insertions(+), 1422 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 Documentation/devicetree/bindings/sound/da7213.txt create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h create mode 100644 sound/soc/codecs/ak4613.c create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h delete mode 100644 sound/soc/codecs/hdmi.c create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h create mode 100644 sound/soc/intel/boards/skl_rt286.c create mode 100644 sound/soc/intel/skylake/skl-topology.c create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h rename {include/sound => sound/soc/sh/rcar}/rcar_snd.h (98%) create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c

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