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

[alsa-devel] [PATCH 0/4] ASoC: rsnd: tidyup CMD (DVC/MIX) setting order
by Kuninori Morimoto 04 Nov '15

04 Nov '15

Hi Mark These are tidyup patches for CMD (= DVC/MIX) for Renesas sound. Current driver settings was not good match with datasheet These patches tidyup settings order Kuninori Morimoto (4): ASoC: rsnd: call clk_prepare()/clk_enable() for AUDIO_CLKx ASoC: rsnd: move CMD related operation to cmd.c ASoC: rsnd: DVC settings matches to datasheet ASoC: rsnd: MIX settings matches to datasheet sound/soc/sh/rcar/adg.c | 5 ++++- sound/soc/sh/rcar/cmd.c | 20 ++++++++++++++++++++ sound/soc/sh/rcar/dvc.c | 141 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++-------------------------------------------------------------- sound/soc/sh/rcar/mix.c | 45 ++++++++++++++++++++++++++------------------ 4 files changed, 128 insertions(+), 83 deletions(-)

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[alsa-devel] [PATCH v1 0/5] Support more sample rates, let the rates normal work
by Caesar Wang 04 Nov '15

04 Nov '15

That's seem the moment rockchip-codec driver other than 44100 and 48000 do not work. Says the (max98090, rt5640). ---- As the chromeos reported issue:(https://chromium.googlesource.com/chromiumos/third_party/kernel/+/v3… For detail: In that bug we want to add 32000 to sound/soc/rockchip/rockchip_max98090.c rk_aif1_hw_params. However that does not work. Previously, in https://chromium-review.googlesource.com/#/c/268398/ , rate 8000, 1600, 96000 were added. However, those rates do not work. I got the same audio as 48000 when I run aplay -D hw:0,0 /usr/local/autotest/cros/audio/fix_440_16_half.raw --rate 8000 -f S16_LE -c2 aplay -D hw:0,0 /usr/local/autotest/cros/audio/fix_440_16_half.raw --rate 16000 -f S16_LE -c2 aplay -D hw:0,0 /usr/local/autotest/cros/audio/fix_440_16_half.raw --rate 96000 -f S16_LE -c2 They all sound identical to aplay -D hw:0,0 /usr/local/autotest/cros/audio/fix_440_16_half.raw --rate 48000 -f S16_LE -c2 Also, recording using aplay -D hw:0,0 /usr/local/test.raw --rate 96000 -f S16_LE -c2 and examine the output we see it was recording using 48000. Okay, let's add this series patchs to fix this issue. arecord -D hw:0 -f S16_LE -c 2 --rate 96000 -d 5 foobar.wav and aplay -D hw:0 foobar.wav Tested on veyron devices. Changes in v1: - change the subject and commit. - remove the print message dev_dbg(). - Fix a subject missspell. Caesar Wang (5): ASoC: rockchip: i2s: Support to set the divider clock API ASoC: rockchip-max98090: Allow more sample rates ASoC: rockchip-rt5645: Allow more sample rates ASoC: rockchip-max98090: Add the divider clock for cpu ASoC: rockchip-rt5645: Add the divider clock for cpu sound/soc/rockchip/rockchip_i2s.c | 27 +++++++++++++++++++++++++ sound/soc/rockchip/rockchip_max98090.c | 37 ++++++++++++++++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 37 ++++++++++++++++++++++++++++++++++ 3 files changed, 101 insertions(+) -- 1.9.1

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[alsa-devel] [PATCH v4 1/2] ASoC: wm9713: convert to regmap
by Robert Jarzmik 04 Nov '15

04 Nov '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. This was tested on the mioa701 board, pxa27x based, in PCM playback, and through suspend/resume. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: fix suspend/resume (that specific part is not tested yet) Since v2: split out the snd_soc_*() from regmap support Since v3: remove suspend/resume regmap cache operations, rely on regmap volatile registers --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9713.c | 174 ++++++++++++++++++++++++++++------------------ 2 files changed, 108 insertions(+), 67 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..22985c42764c 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> @@ -39,33 +40,15 @@ 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, -}; +static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + return snd_soc_read(codec, reg); +} +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + return snd_soc_write(codec, reg, val); +} #define HPL_MIXER 0 #define HPR_MIXER 1 @@ -674,39 +657,97 @@ 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 { @@ -1173,8 +1214,7 @@ 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 +1229,8 @@ 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_mark_dirty(codec->component.regmap); + snd_soc_cache_sync(codec); } return ret; @@ -1203,6 +1239,7 @@ 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); + struct regmap *regmap; int reg; wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, @@ -1210,6 +1247,14 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) 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); @@ -1221,6 +1266,7 @@ static int wm9713_soc_remove(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9713->ac97); return 0; } @@ -1230,13 +1276,7 @@ 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] [PATCH 1/3] ASoC: rockchip: i2s: Add set the divider clock API
by Caesar Wang 04 Nov '15

04 Nov '15

In order to support more rates, add the divider clock api. As the input source clock to the module is MCLK_I2S, and by the divider of the module, the clock generator generates SCLK and LRCK to transmitter and receiver. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index b936102..2086a6a 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -286,6 +286,34 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, return ret; } +static int rockchip_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct rk_i2s_dev *i2s = to_info(cpu_dai); + unsigned int val = 0; + + dev_dbg(i2s->dev, "%s: div_id=%d, div=%d\n", __func__, div_id, div); + + switch (div_id) { + case ROCKCHIP_DIV_BCLK: + val |= I2S_CKR_TSD(div); + val |= I2S_CKR_RSD(div); + regmap_update_bits(i2s->regmap, I2S_CKR, + I2S_CKR_TSD_MASK | I2S_CKR_RSD_MASK, + val); + break; + case ROCKCHIP_DIV_MCLK: + val |= I2S_CKR_MDIV(div); + regmap_update_bits(i2s->regmap, I2S_CKR, + I2S_CKR_MDIV_MASK, val); + break; + default: + return -EINVAL; + } + + return 0; +} + static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -311,6 +339,7 @@ static int rockchip_i2s_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { .hw_params = rockchip_i2s_hw_params, + .set_clkdiv = rockchip_i2s_set_clkdiv, .set_sysclk = rockchip_i2s_set_sysclk, .set_fmt = rockchip_i2s_set_fmt, .trigger = rockchip_i2s_trigger, -- 1.9.1

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Re: [alsa-devel] [PATCH] Intel device IDs Audio
by Takashi Iwai 04 Nov '15

04 Nov '15

On Wed, 04 Nov 2015 01:12:47 +0100, Alexandra Yates wrote: > > Adding Intel codename Lewisburg platform device IDs for audio. > > Signed-off-by: Alexandra Yates <alexandra.yates(a)linux.intel.com> > --- > sound/pci/hda/hda_intel.c | 5 +++++ > 1 file changed, 5 insertions(+) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index c38c68f..51d6ff3 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -2099,6 +2099,11 @@ static const struct pci_device_id azx_ids[] = { > /* 9 Series */ > { PCI_DEVICE(0x8086, 0x8ca0), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > + /* Lewisburg */ > + { PCI_DEVICE(0x8086, 0xa1f0), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > + { PCI_DEVICE(0x8086, 0xa270), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > /* Wellsburg */ > { PCI_DEVICE(0x8086, 0x8d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, Is this the best position to add? That is, is Lewisburg an older chip than Wellsburg? thanks, Takashi

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[alsa-devel] [PATCH] ASoC: wm8998: Remove duplicated consts
by Richard Fitzgerald 03 Nov '15

03 Nov '15

The SOC_xxx_DECL() macros already include 'const' so there's no need to put a const in the source where they are used. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8998.c | 46 +++++++++++++++++++++++----------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 8782dfb..7719bc5 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -199,20 +199,20 @@ static const char * const wm8998_inmux_texts[] = { "B", }; -static const SOC_ENUM_SINGLE_DECL(wm8998_in1muxl_enum, - ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_IN1L_SRC_SHIFT, - wm8998_inmux_texts); +static SOC_ENUM_SINGLE_DECL(wm8998_in1muxl_enum, + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_SRC_SHIFT, + wm8998_inmux_texts); -static const SOC_ENUM_SINGLE_DECL(wm8998_in1muxr_enum, - ARIZONA_ADC_DIGITAL_VOLUME_1R, - ARIZONA_IN1R_SRC_SHIFT, - wm8998_inmux_texts); +static SOC_ENUM_SINGLE_DECL(wm8998_in1muxr_enum, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_SRC_SHIFT, + wm8998_inmux_texts); -static const SOC_ENUM_SINGLE_DECL(wm8998_in2mux_enum, - ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_IN2L_SRC_SHIFT, - wm8998_inmux_texts); +static SOC_ENUM_SINGLE_DECL(wm8998_in2mux_enum, + ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_SRC_SHIFT, + wm8998_inmux_texts); static const struct snd_kcontrol_new wm8998_in1mux[2] = { SOC_DAPM_ENUM_EXT("IN1L Mux", wm8998_in1muxl_enum, @@ -522,17 +522,17 @@ static const unsigned int wm8998_aec_loopback_values[] = { 0, 1, 2, 3, 4, 6, 7, 8, 9, }; -static const SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec1_loopback, - ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, - wm8998_aec_loopback_texts, - wm8998_aec_loopback_values); - -static const SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec2_loopback, - ARIZONA_DAC_AEC_CONTROL_2, - ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, - wm8998_aec_loopback_texts, - wm8998_aec_loopback_values); +static SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec1_loopback, + ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + wm8998_aec_loopback_texts, + wm8998_aec_loopback_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec2_loopback, + ARIZONA_DAC_AEC_CONTROL_2, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + wm8998_aec_loopback_texts, + wm8998_aec_loopback_values); static const struct snd_kcontrol_new wm8998_aec_loopback_mux[] = { SOC_DAPM_ENUM("AEC1 Loopback", wm8998_aec1_loopback), -- 1.9.1

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[alsa-devel] [PATCH] ASoC: arizona: fix range of OPCLK_REF
by Richard Fitzgerald 03 Nov '15

03 Nov '15

The code was able to generate illegal OPCLK_REF values because the reference frequency tables listed all values of SYSCLK instead of valid values for OPCLK_REF clock. The maximum OPCLK_REF clock is 49.152MHz or 45.1584MHz. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9929efc..b3ea24d 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1023,24 +1023,18 @@ void arizona_init_dvfs(struct arizona_priv *priv) } EXPORT_SYMBOL_GPL(arizona_init_dvfs); -static unsigned int arizona_sysclk_48k_rates[] = { +static unsigned int arizona_opclk_ref_48k_rates[] = { 6144000, 12288000, 24576000, 49152000, - 73728000, - 98304000, - 147456000, }; -static unsigned int arizona_sysclk_44k1_rates[] = { +static unsigned int arizona_opclk_ref_44k1_rates[] = { 5644800, 11289600, 22579200, 45158400, - 67737600, - 90316800, - 135475200, }; static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, @@ -1065,11 +1059,11 @@ static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, } if (refclk % 8000) - rates = arizona_sysclk_44k1_rates; + rates = arizona_opclk_ref_44k1_rates; else - rates = arizona_sysclk_48k_rates; + rates = arizona_opclk_ref_48k_rates; - for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) && + for (ref = 0; ref < ARRAY_SIZE(arizona_opclk_ref_48k_rates) && rates[ref] <= refclk; ref++) { div = 1; while (rates[ref] / div >= freq && div < 32) { -- 1.9.1

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[alsa-devel] Applied "ASoC: wm8960: Fix the Input PGA Mute switch" to the asoc tree
by Mark Brown 03 Nov '15

03 Nov '15

The patch ASoC: wm8960: Fix the Input PGA Mute switch has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 41a59cae585678136c28cdcbba9cb2faf27685f5 Mon Sep 17 00:00:00 2001 From: JongHo Kim <furmuwon(a)gmail.com> Date: Tue, 3 Nov 2015 11:06:32 +0900 Subject: [PATCH] ASoC: wm8960: Fix the Input PGA Mute switch Change the xinvert value from 0 to 1 on the "Capture Switch" control WM8960 datasheet is shown as follows: Bit 7 at 00h and 01h register address 1 : Enable Mute, 0 : Disable Mute Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e3b7d0c57411..bbe24275f8c9 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -223,7 +223,7 @@ SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, 6, 1, 0), SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume", WM8960_INBMIX1, 4, 7, 0, boost_tlv), -- 2.6.2

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[alsa-devel] [PATCH MC Next Gen v2 0/3] Add mixer and control interface media entities
by Shuah Khan 03 Nov '15

03 Nov '15

This patch v2 series adds the following: A fix to media_stream_delete() to remove intf devnode. Add support for creating MEDIA_ENT_F_AUDIO_MIXER entity for each mixer and a MEDIA_INTF_T_ALSA_CONTROL control interface entity that links to mixer entities. MEDIA_INTF_T_ALSA_CONTROL entity corresponds to the control device for the card. Change au0828_create_media_graph() to create pad link between MEDIA_ENT_F_AUDIO_MIXER entity and decoder's AU8522_PAD_AUDIO_OUT. With mixer entity now linked to decoder, change to link MEDIA_ENT_F_AUDIO_CAPTURE to mixer's source pad Changes since v1: 1. Included the fix to media_stream_delete() which was by mistake not included in the patch v1. 2. Fixed mixer to decoder link and mider to capture node links based on Mauro Chehapb's review comments on the media graph. This patch series is dependent on an earlier patch series: Update ALSA, and au0828 drivers to use Managed Media Controller API: https://www.mail-archive.com/linux-media@vger.kernel.org/msg92752.html Please find the media graphs for ALSA work with mixer and control interface nodes: https://drive.google.com/folderview?id=0B0NIL0BQg-AlLWE3SzAxazBJWm8&usp=sha… Shuah Khan (3): sound/usb: Fix media_stream_delete() to remove intf devnode sound/usb: Create media mixer function and control interface entities media: au0828 create link between ALSA Mixer and decoder drivers/media/usb/au0828/au0828-core.c | 17 +++++-- drivers/media/usb/au0828/au0828.h | 1 + sound/usb/card.c | 5 ++ sound/usb/media.c | 90 ++++++++++++++++++++++++++++++++++ sound/usb/media.h | 20 ++++++++ sound/usb/mixer.h | 1 + sound/usb/usbaudio.h | 1 + 7 files changed, 132 insertions(+), 3 deletions(-) -- 2.1.4

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[alsa-devel] [RFT][PATCH 1/2] ASoC: rt286: Fix run time error while modifying const data
by Axel Lin 03 Nov '15

03 Nov '15

Make a copy of memory for index_cache rather than directly use the rt286_index_def to avoid run time error. Fixes: c418a84a8c8f ("ASoC: Constify reg_default tables") Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt286.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1fbdb4f..af2ed77 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -37,7 +37,7 @@ #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { - const struct reg_default *index_cache; + struct reg_default *index_cache; int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; @@ -1160,7 +1160,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - rt286->index_cache = rt286_index_def; + rt286->index_cache = devm_kmemdup(&i2c->dev, rt286_index_def, + sizeof(rt286_index_def), GFP_KERNEL); + if (!rt286->index_cache) + return -ENOMEM; + rt286->index_cache_size = INDEX_CACHE_SIZE; rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); -- 2.1.4

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