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

[alsa-devel] [PATCH v3 5/8] ASoC: da7210: Add support for ADC & DAC equalizers
by Ashish Chavan 15 Oct '11

15 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 bc40fd8..8dad77e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -40,10 +40,16 @@ #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 @@ -191,6 +197,8 @@ 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), @@ -241,6 +249,34 @@ 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), }; /* -- 1.7.1

2 2

[alsa-devel] [PATCH v3 1/8] ASoC: da7210: bugfix for head phone volume control
by Ashish Chavan 14 Oct '11

14 Oct '11

This patch takes care of reserved bits of headphone volume register by using correct volume range. 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 | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ff68247..0809f32 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -149,12 +149,14 @@ * 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 struct snd_kcontrol_new da7210_snd_controls[] = { -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: wm8990: Use snd_soc_update_bits for read-modify-write
by Axel Lin 14 Oct '11

14 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write This patch also includes a comment fix in wm8990_set_dai_pll(), if freq_in and freq_out are 0, what we do is to clear WM8990_PLL_ENA bit. Thus the comment should be "Turn off PLL". Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8990.c | 67 ++++++++++++++++++--------------------------- 1 files changed, 27 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index b9c5ecc..d29a962 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -981,7 +981,6 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - u16 reg; struct snd_soc_codec *codec = codec_dai->codec; struct _pll_div pll_div; @@ -989,13 +988,12 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, pll_factors(&pll_div, freq_out * 4, freq_in); /* Turn on PLL */ - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - reg |= WM8990_PLL_ENA; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_PLL_ENA, WM8990_PLL_ENA); /* sysclk comes from PLL */ - reg = snd_soc_read(codec, WM8990_CLOCKING_2); - snd_soc_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_SYSCLK_SRC, WM8990_SYSCLK_SRC); /* set up N , fractional mode and pre-divisor if necessary */ snd_soc_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | @@ -1003,10 +1001,9 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); snd_soc_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); } else { - /* Turn on PLL */ - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - reg &= ~WM8990_PLL_ENA; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_PLL_ENA, 0); } return 0; } @@ -1084,28 +1081,23 @@ static int wm8990_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 WM8990_MCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_MCLK_DIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_MCLK_DIV_MASK, div); break; case WM8990_DACCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_DAC_CLKDIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_DAC_CLKDIV_MASK, div); break; case WM8990_ADCCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_2) & - ~WM8990_ADC_CLKDIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_2, + WM8990_ADC_CLKDIV_MASK, div); break; case WM8990_BCLK_DIV: - reg = snd_soc_read(codec, WM8990_CLOCKING_1) & - ~WM8990_BCLK_DIV_MASK; - snd_soc_write(codec, WM8990_CLOCKING_1, reg | div); + snd_soc_update_bits(codec, WM8990_CLOCKING_1, + WM8990_BCLK_DIV_MASK, div); break; default: return -EINVAL; @@ -1164,7 +1156,6 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { int ret; - u16 val; switch (level) { case SND_SOC_BIAS_ON: @@ -1172,9 +1163,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & - ~WM8990_VMID_MODE_MASK; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1, + WM8990_VMID_MODE_MASK, 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1239,9 +1229,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, } /* VMID=2*250k */ - val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & - ~WM8990_VMID_MODE_MASK; - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1, + WM8990_VMID_MODE_MASK, 0x4); break; case SND_SOC_BIAS_OFF: @@ -1255,8 +1244,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, WM8990_BUFIOEN); /* mute DAC */ - val = snd_soc_read(codec, WM8990_DAC_CTRL); - snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + snd_soc_update_bits(codec, WM8990_DAC_CTRL, + WM8990_DAC_MUTE, WM8990_DAC_MUTE); /* Enable any disabled outputs */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); @@ -1344,7 +1333,6 @@ static int wm8990_resume(struct snd_soc_codec *codec) static int wm8990_probe(struct snd_soc_codec *codec) { int ret; - u16 reg; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret < 0) { @@ -1357,15 +1345,14 @@ static int wm8990_probe(struct snd_soc_codec *codec) /* charge output caps */ wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4); - snd_soc_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); + snd_soc_update_bits(codec, WM8990_AUDIO_INTERFACE_4, + WM8990_ALRCGPIO1, WM8990_ALRCGPIO1); - reg = snd_soc_read(codec, WM8990_GPIO1_GPIO2) & - ~WM8990_GPIO1_SEL_MASK; - snd_soc_write(codec, WM8990_GPIO1_GPIO2, reg | 1); + snd_soc_update_bits(codec, WM8990_GPIO1_GPIO2, + WM8990_GPIO1_SEL_MASK, 1); - reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); - snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); + snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2, + WM8990_OPCLK_ENA, WM8990_OPCLK_ENA); snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8400: Fix wrong bit setting for WM8400_POWER_MANAGEMENT_2
by Axel Lin 14 Oct '11

14 Oct '11

If (fakepower & ((1 << WM8400_INMIXR_PWR) | (1 << WM8400_AINRMUX_PWR))) is false, we should clear WM8400_AINR_ENA bits instead of WM8400_AINL_ENA. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8400.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index fbee556..dc13be2 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -383,7 +383,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, (1 << WM8400_AINRMUX_PWR))) { reg |= WM8400_AINR_ENA; } else { - reg &= ~WM8400_AINL_ENA; + reg &= ~WM8400_AINR_ENA; } wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); -- 1.7.4.1

2 2

[alsa-devel] [PATCH 1/2] ASoC: wm8991: Fix wrong bit setting for WM8991_POWER_MANAGEMENT_2
by Axel Lin 14 Oct '11

14 Oct '11

If (fakepower & ((1 << WM8991_INMIXR_PWR_BIT)|(1 << WM8991_AINRMUX_PWR_BIT)))) is false, we should clear WM8991_AINR_ENA bits instead of WM8991_AINL_ENA. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8991.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 08d64a6..708d251 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -393,7 +393,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w, (1 << WM8991_AINRMUX_PWR_BIT))) reg |= WM8991_AINR_ENA; else - reg &= ~WM8991_AINL_ENA; + reg &= ~WM8991_AINR_ENA; snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); return 0; -- 1.7.4.1

2 2

[alsa-devel] [PATCH 1/3] ASoC: alc5623: Convert codec->hw_read to snd_soc_read
by Axel Lin 14 Oct '11

14 Oct '11

codec->hw_read is broken now, let's covert to snd_soc_read. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 557b3af..984b14b 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -53,8 +53,10 @@ static void alc5623_fill_cache(struct snd_soc_codec *codec) u16 *cache = codec->reg_cache; /* not really efficient ... */ + codec->cache_bypass = 1; for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = codec->hw_read(codec, i); + cache[i] = snd_soc_read(codec, i); + codec->cache_bypass = 0; } static inline int alc5623_reset(struct snd_soc_codec *codec) -- 1.7.4.1

2 7

[alsa-devel] [PATCH v2] ASoC: tlv320aic23: convert to soc-cache
by Axel Lin 14 Oct '11

14 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: Use snd_soc_cache_write instead of snd_soc_write with cache_only flag to write the register default value to cache for reserved registers. sound/soc/codecs/tlv320aic23.c | 167 ++++++++++++--------------------------- 1 files changed, 52 insertions(+), 115 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3a4bb2..ab27dbc 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -47,63 +47,6 @@ static const u16 tlv320aic23_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ }; -/* - * read tlv320aic23 register cache - */ -static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec - *codec, unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return -1; - return cache[reg]; -} - -/* - * write tlv320aic23 register cache - */ -static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, - u8 reg, u16 value) -{ - u16 *cache = codec->reg_cache; - if (reg >= ARRAY_SIZE(tlv320aic23_reg)) - return; - cache[reg] = value; -} - -/* - * write to the tlv320aic23 register space - */ -static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - - u8 data[2]; - - /* TLV320AIC23 has 7 bit address and 9 bits of data - * so we need to switch one data bit into reg and rest - * of data into val - */ - - if (reg > 9 && reg != 15) { - printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); - return -1; - } - - data[0] = (reg << 1) | (value >> 8 & 0x01); - data[1] = value & 0xff; - - tlv320aic23_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - - printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, - value, reg); - - return -EIO; -} - static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -139,8 +82,8 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, */ val = (val >= 4) ? 4 : (3 - val); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); - tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0); + snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); return 0; } @@ -151,7 +94,7 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u16 val; - val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0); val = val >> 6; val = (val >= 4) ? 4 : (3 - val); ucontrol->value.integer.value[0] = val; @@ -232,7 +175,6 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { enum snd_soc_control_type control_type; - void *control_data; int mclk; int requested_adc; int requested_dac; @@ -344,7 +286,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int src = snd_soc_read(codec, TLV320AIC23_SRATE); int sr = (src >> 2) & 0x0f; int val = (mclk / bosr_usb_divisor_table[src & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; @@ -368,7 +310,7 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, __func__, sample_rate_adc, sample_rate_dac); return -EINVAL; } - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + snd_soc_write(codec, TLV320AIC23_SRATE, data); #ifdef DEBUG { u32 adc, dac; @@ -407,9 +349,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - iface_reg = - tlv320aic23_read_reg_cache(codec, - TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -423,7 +364,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg |= (0x03 << 2); break; } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -435,7 +376,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; /* set active */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); return 0; } @@ -450,7 +391,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aic23->requested_dac = 0; @@ -463,14 +404,14 @@ static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; u16 reg; - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + reg = snd_soc_read(codec, TLV320AIC23_DIGT); if (mute) reg |= TLV320AIC23_DACM_MUTE; else reg &= ~TLV320AIC23_DACM_MUTE; - tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + snd_soc_write(codec, TLV320AIC23_DIGT, reg); return 0; } @@ -481,8 +422,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 iface_reg; - iface_reg = - tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -516,7 +456,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, } - tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); return 0; } @@ -532,26 +472,26 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ TLV320AIC23_DAC_OFF); - tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + snd_soc_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \ - TLV320AIC23_CLK_OFF); + snd_soc_write(codec, TLV320AIC23_PWR, + reg | TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); - tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); + snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); break; } codec->dapm.bias_level = level; @@ -598,13 +538,7 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec, static int tlv320aic23_resume(struct snd_soc_codec *codec) { - u16 reg; - - /* Sync reg_cache with the hardware */ - for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { - u16 val = tlv320aic23_read_reg_cache(codec, reg); - tlv320aic23_write(codec, reg, val); - } + snd_soc_cache_sync(codec); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -613,46 +547,52 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); - int reg; + int ret; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec->control_data = aic23->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; - codec->hw_read = NULL; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* Reset codec */ - tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + snd_soc_write(codec, TLV320AIC23_RESET, 0); + + /* Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + snd_soc_cache_write(codec, 0x0A, 0); + snd_soc_cache_write(codec, 0x0B, 0); + snd_soc_cache_write(codec, 0x0C, 0); + snd_soc_cache_write(codec, 0x0D, 0); + snd_soc_cache_write(codec, 0x0E, 0); /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); - tlv320aic23_write(codec, TLV320AIC23_LINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - (TLV320AIC23_LRS_ENABLED)); + snd_soc_update_bits(codec, TLV320AIC23_LINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); - tlv320aic23_write(codec, TLV320AIC23_RINVOL, - (reg & (~TLV320AIC23_LIM_MUTED)) | - TLV320AIC23_LRS_ENABLED); + snd_soc_update_bits(codec, TLV320AIC23_RINVOL, + TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED); - reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); - tlv320aic23_write(codec, TLV320AIC23_ANLG, - (reg) & (~TLV320AIC23_BYPASS_ON) & - (~TLV320AIC23_MICM_MUTED)); + snd_soc_update_bits(codec, TLV320AIC23_ANLG, + TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED, + 0); /* Default output volume */ - tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, - TLV320AIC23_DEFAULT_OUT_VOL & - TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); + snd_soc_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK); - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); @@ -674,8 +614,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, - .read = tlv320aic23_read_reg_cache, - .write = tlv320aic23_write, .set_bias_level = tlv320aic23_set_bias_level, .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), @@ -702,7 +640,6 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, aic23); - aic23->control_data = i2c; aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

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[alsa-devel] [PATCH v3 3/8] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 14 Oct '11

14 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 b6e5085..a7ea07d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -42,6 +42,8 @@ #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 @@ -141,6 +143,17 @@ #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" /* @@ -161,11 +174,35 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +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), }; /* @@ -659,6 +696,12 @@ 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: max98095: Convert codec->hw_write to snd_soc_write
by Axel Lin 14 Oct '11

14 Oct '11

codec->hw_write is broken now, convert codec->hw_write to snd_soc_write. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi, I don't have this hardware, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/max98095.c | 26 ++++---------------------- 1 files changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 6982f74..85f9379 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -610,24 +610,6 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) } /* - * Filter coefficients are in a separate register segment - * and they share the address space of the normal registers. - * The coefficient registers do not need or share the cache. - */ -static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - data[0] = reg; - data[1] = value; - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -/* * Load equalizer DSP coefficient configurations registers */ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, @@ -647,8 +629,8 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); } } @@ -672,8 +654,8 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); } } -- 1.7.4.1

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[alsa-devel] [PATCH 0/2] Conexant HD-audio fixes
by Takashi Iwai 14 Oct '11

14 Oct '11

Hi, while debugging a problem reported by a Thinkpad user, we found that there are fundamental problems in the current Conexant auto-parser code. 1. EAPDs on older chips don't correspond to the pin as 1:1. A single EAPD can influence on the whole outputs, so we can't turn it off simply even when the assigned pin isn't used. 2. There is only a single input-amp in audio-in widgets. This is contradicting against what I was informed before. There is only one capture level in each audio-in widget no matter how many sources are connected to it. For example, if an ADC has the connection to an internal mic and an external mic, the volume in ADC is only one, no individual amp per connection, unlike other widgets. For the first issue, a fix is just keep EAPD turned on for such chips. This simplifies the previous David's fix. About the second part, I'm not 100% sure whether it's correct to all Conexant chips. But this sounds more natural, and I always wondered about the old implementation. Can anyone confirm that the patches work properly? The initial test was done with a machine with cx20549, so tests with other codecs like cx20551, cx20561, and cx2058x would be appreciated. After applying the patch, load snd-hda-intel with model=auto option. You'll have a single "Capture" volume (or two capture volumes like "Mic" and "Internal Mic", which correspond to different ADCs). Check the different recording source and whether the capture volume(s) really changes the recording level properly. thanks, Takashi

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