October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 3/9] ASoC: da7210: Add support for ADC & DAC equalizers
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds support for ADC and DAC five band equalizers available on DA7210 codec. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 87b9ae5..7e36ed6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -38,10 +38,16 @@ #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D #define DA7210_OUT1_L 0x1E @@ -181,6 +187,8 @@ static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -231,6 +239,34 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, inpga_gain_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), }; /* Codec private data */ -- 1.7.1

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[alsa-devel] [PATCH v2 1/9] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 13 Oct '11

13 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Removed explicit setting of default gains - Renamed "DAC Playback" to "Digital Playback" --- sound/soc/codecs/da7210.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 43 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a9d9d39..1ef058b 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -40,6 +40,9 @@ #define DA7210_DAC_SEL 0x17 #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -133,6 +136,17 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -150,12 +164,35 @@ * -> min = -79.5 dB */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); + +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), }; /* Codec private data */ @@ -478,6 +515,12 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] [PATCH] ASoC: ak4535: fixup cache register table
by Axel Lin 13 Oct '11

13 Oct '11

ak4535_reg should be 8bit, but cache table is defined as 16bit. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4535.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 68df32d..95d782d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -39,11 +39,11 @@ struct ak4535_priv { /* * ak4535 register cache */ -static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { - 0x0000, 0x0080, 0x0000, 0x0003, - 0x0002, 0x0000, 0x0011, 0x0001, - 0x0000, 0x0040, 0x0036, 0x0010, - 0x0000, 0x0000, 0x0057, 0x0000, +static const u8 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x00, 0x80, 0x00, 0x03, + 0x02, 0x00, 0x11, 0x01, + 0x00, 0x40, 0x36, 0x10, + 0x00, 0x00, 0x57, 0x00, }; static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: ak4642: convert to soc-cache
by Axel Lin 13 Oct '11

13 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi Kuninori, This patch converts ak4642 to use soc-cache APIs. I don't have this hardware, I'd appreciate if you can test this patch. Thanks, Axel sound/soc/codecs/ak4642.c | 82 ++++++-------------------------------------- 1 files changed, 12 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 68fc16c..80a9fe8 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -157,7 +157,6 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { struct ak4642_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -176,64 +175,6 @@ static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { 0x0000, }; -/* - * read ak4642 register cache - */ -static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4642_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4642 register cache - */ -static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4642_CACHEREGNUM) - return; - - cache[reg] = value; -} - -/* - * write to the AK4642 register space - */ -static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4642 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (codec->hw_write(codec->control_data, data, 2) == 2) { - ak4642_write_reg_cache(codec, reg, value); - return 0; - } else - return -EIO; -} - -static int ak4642_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4642_CACHEREGNUM; i++) - r |= ak4642_write(codec, i, cache[i]); - - return r; -}; - static int ak4642_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -253,8 +194,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, */ snd_soc_update_bits(codec, MD_CTL4, DACH, DACH); snd_soc_update_bits(codec, MD_CTL3, BST1, BST1); - ak4642_write(codec, L_IVC, 0x91); /* volume */ - ak4642_write(codec, R_IVC, 0x91); /* volume */ + snd_soc_write(codec, L_IVC, 0x91); /* volume */ + snd_soc_write(codec, R_IVC, 0x91); /* volume */ snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC, PMVCM | PMMIN | PMDAC); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); @@ -273,9 +214,9 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - ak4642_write(codec, SG_SL1, PMMP | MGAIN0); - ak4642_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); - ak4642_write(codec, ALC_CTL1, ALC | LMTH0); + snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); + snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMADL, PMVCM | PMADL); snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); @@ -463,7 +404,7 @@ static struct snd_soc_dai_driver ak4642_dai = { static int ak4642_resume(struct snd_soc_codec *codec) { - ak4642_sync(codec); + snd_soc_cache_sync(codec); return 0; } @@ -471,11 +412,15 @@ static int ak4642_resume(struct snd_soc_codec *codec) static int ak4642_probe(struct snd_soc_codec *codec) { struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION); - codec->hw_write = (hw_write_t)i2c_master_send; - codec->control_data = ak4642->control_data; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } snd_soc_add_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); @@ -486,8 +431,6 @@ static int ak4642_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .probe = ak4642_probe, .resume = ak4642_resume, - .read = ak4642_read_reg_cache, - .write = ak4642_write, .reg_cache_size = ARRAY_SIZE(ak4642_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4642_reg, @@ -505,7 +448,6 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4642); - ak4642->control_data = i2c; ak4642->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

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[alsa-devel] [RFC][PATCH] ASoC: Suppress restore of default register values for lzo cache sync
by Axel Lin 13 Oct '11

13 Oct '11

Currently we do suppress restore of default register values for rbtree and flat cache sync. A couple users use this trick to avoid writing reset register while resume. Thus make lzo cache sync suppress restore of default register values, so we have consistent behavior for all types of cache sync. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/soc-cache.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9077aa4..eb3c016 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -555,16 +555,22 @@ static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) { struct snd_soc_lzo_ctx **lzo_blocks; - unsigned int val; + unsigned int val, def; + unsigned int word_size; int i; int ret; lzo_blocks = codec->reg_cache; + word_size = codec->driver->reg_word_size; for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { WARN_ON(!snd_soc_codec_writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; + def = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size); + if (val == def) + continue; + codec->cache_bypass = 1; ret = snd_soc_write(codec, i, val); codec->cache_bypass = 0; -- 1.7.4.1

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 13 Oct '11

13 Oct '11

The following changes since commit 03b5362d2fef02cabbbb440d861d27e40ed3a929: ASoC: Convert Jive to table based init (2011-10-10 11:10:44 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Ashish Chavan (1): ASoC: da7210: Add support for other DAI word lengths, format and mode Axel Lin (6): ASoC: Writing register default value for the reset register ASoC: ak4642: convert to soc-cache ASoC: Delete ads117x.h ASoC: Remove impossible case from wm8994_hw_params ASoC: da7210: convert to soc-cache ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X Mark Brown (4): ASoC: Convert Goni to data based DAPM init ASoC: Add missing default for WM5100 Clocking 1 ASoC: Implement WM5100 accessory detection support ASoC: Instantiate card widgets immediately Peter Ujfalusi (18): ASoC: OMAP machines: Remove soc_dapm_sync() call from init ASoC: zoom2: Let core to deal with the DAPM widgets ASoC: zoom2: No need to call dapm_pin_enable at init time ASoC: sdp4430: Let core to deal with the DAPM widgets ASoC: sdp4430: No need to call dapm_pin_enable at init time ASoC: am3517evm: Let core to deal with the DAPM widgets ASoC: n810: Let the core to register DAPM widgets/routes and controls ASoC: osk5912: Let core to deal with the DAPM widgets ASoC: sdp3430: Let core to deal with the DAPM widgets ASoC: twl6040: Convert to table based init ASoC: twl4030: Convert to table based init ASoC: tlv320dac33: Convert to table based init Input: twl6040: Simplify vibra regsiter definitions MFD: twl6040: Cache the vibra control registers MFD: twl6040: function to query the vibra status for clients Input: twl6040-vibra: Check the selected path for vibra ASoC: twl6040: Support for vibra output paths ASoC: twl6040: Workaround for headset DC offset caused pop noise Wolfram Sang (1): ASoC: Fix typo in Kconfig symbol for tlv320aic32x4 drivers/input/misc/twl6040-vibra.c | 19 +++-- drivers/mfd/twl6040-core.c | 31 ++++++- include/linux/mfd/twl6040.h | 25 +++--- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/ads117x.h | 13 --- sound/soc/codecs/ak4642.c | 82 +++---------------- sound/soc/codecs/da7210.c | 146 +++++++++++++++----------------- sound/soc/codecs/tlv320dac33.c | 23 ++---- sound/soc/codecs/twl4030.c | 21 ++--- sound/soc/codecs/twl6040.c | 135 ++++++++++++++++++++++++------ sound/soc/codecs/wm5100-tables.c | 1 + sound/soc/codecs/wm5100.c | 162 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm5100.h | 2 + sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8994.c | 10 -- sound/soc/imx/Kconfig | 2 +- sound/soc/omap/am3517evm.c | 28 +----- sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/n810.c | 23 ++---- sound/soc/omap/omap3pandora.c | 8 +-- sound/soc/omap/osk5912.c | 27 +----- sound/soc/omap/rx51.c | 2 - sound/soc/omap/sdp3430.c | 18 +--- sound/soc/omap/sdp4430.c | 27 +----- sound/soc/omap/zoom2.c | 26 ++----- sound/soc/pxa/Kconfig | 2 + sound/soc/samsung/goni_wm8994.c | 13 +-- sound/soc/soc-core.c | 2 + 29 files changed, 467 insertions(+), 390 deletions(-) delete mode 100644 sound/soc/codecs/ads117x.h

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[alsa-devel] [PATCH] ASoC: sta32x: Set reg_cache_default to sta32x_regs
by Axel Lin 13 Oct '11

13 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sta32x.c | 5 +---- 1 files changed, 1 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 5c7def3..754b3ff 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -756,10 +756,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) return ret; } - /* read reg reset values into cache */ - for (i = 0; i < STA32X_REGISTER_COUNT; i++) - snd_soc_cache_write(codec, i, sta32x_regs[i]); - /* preserve reset values of reserved register bits */ snd_soc_cache_write(codec, STA32X_CONFC, codec->hw_read(codec, STA32X_CONFC)); @@ -837,6 +833,7 @@ static const struct snd_soc_codec_driver sta32x_codec = { .resume = sta32x_resume, .reg_cache_size = STA32X_REGISTER_COUNT, .reg_word_size = sizeof(u8), + .reg_cache_default = sta32x_regs, .volatile_register = sta32x_reg_is_volatile, .set_bias_level = sta32x_set_bias_level, .controls = sta32x_snd_controls, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: pxa: Remove redundant snd_soc_dapm_sync() calls from machine drivers
by Axel Lin 13 Oct '11

13 Oct '11

The core will sync DAPM as part of the card initialization, there is no need for machine drivers to do so during their setup. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/poodle.c | 1 - sound/soc/pxa/spitz.c | 1 - 2 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index da3ae43..4c29bc1 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -265,7 +265,6 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd) /* Set up poodle specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index ce920e3..c2d6ff9 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -301,7 +301,6 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) /* Set up spitz specific audio paths */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(dapm); return 0; } -- 1.7.4.1

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[alsa-devel] [PATCH 7/9] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 104 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 104 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index eb12c82..9cab62a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,6 +26,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 @@ -65,6 +66,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -87,6 +94,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -165,9 +173,11 @@ #define DA7210_OUT2_EN (1 << 7) /* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) #define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) #define DA7210_AUX1_R_EN (1 << 7) /* AUX2 bit fields */ @@ -176,6 +186,19 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + /* Default gain/vol values */ #define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ #define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ @@ -185,6 +208,10 @@ #define DA7210_DFLT_AUX2_VOL 0x02 /* 6dB */ #define DA7210_DFLT_INPGA_VOL 0x07 /* 6dB */ +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -247,6 +274,70 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 *reg_cache = codec->reg_cache; + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (reg_cache[DA7210_CONTROL] & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, "Failed to enable ALC\n"); + return -EPERM; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 *reg_cache = codec->reg_cache; + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (reg_cache[DA7210_ADC] & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((reg_cache[DA7210_ZERO_CROSS] & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != 0xc3) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = reg_cache[DA7210_IN_GAIN]; + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((reg_cache[DA7210_AUX1_L] & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((reg_cache[DA7210_AUX1_R] & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + dev_dbg(codec->dev, "Failed to enable Noise suppression\n"); + return -EPERM; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -322,6 +413,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Gain", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Gain", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Gain", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- 1.7.1

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[alsa-devel] [PATCH 6/9] ASoC: da7210: bugfix for head phone volume control
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch takes care of reserved bits of headphone volume register by using correct volume range. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 60f403b..eb12c82 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -196,12 +196,15 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); -- 1.7.1

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