February 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: fix WM8753 initialization.
by Denis 'GNUtoo' Carikli 04 Mar '12

04 Mar '12

Without that fix the wm8753 SPI initialization fails, and then produces a kernel panic during boot with the following call trace: Unable to handle kernel paging request at virtual address 37386d9b [<c01ccafc>] (regmap_get_val_bytes+0x0/0x14) from [<c0243dfc>] (snd_soc_codec_set_cache_io+0x9c/0xcc) [<c0243dfc>] (snd_soc_codec_set_cache_io+0x9c/0xcc) from [<c0244a4c>] (wm8753_probe+0x5c/0x1c4) [<c0244a4c>] (wm8753_probe+0x5c/0x1c4) from [<c023bb24>] (soc_probe_codec+0x174/0x284) [<c023bb24>] (soc_probe_codec+0x174/0x284) from [<c023c2c0>] (snd_soc_instantiate_cards+0x68c/0xe28) [<c023c2c0>] (snd_soc_instantiate_cards+0x68c/0xe28) from [<c023d278>] (snd_soc_register_card+0x240/0x2d4) [<c023d278>] (snd_soc_register_card+0x240/0x2d4) from [<c023d330>] (soc_probe+0x24/0x40) [<c023d330>] (soc_probe+0x24/0x40) from [<c01c3900>] (platform_drv_probe+0x14/0x18) [...] The commit d3398ff05907167f463e119421b053ce043741d1 ( ASoC: Convert WM8753 to direct regmap API usage ) introduced the problem. Thanks to Lars-Peter Clausen for helping me a bit during the debugging. Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo(a)no-log.org> --- sound/soc/codecs/wm8753.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 21ed75d..98fb921 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1577,6 +1577,9 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); goto err_regmap; } + + return 0; + err_regmap: regmap_exit(wm8753->regmap); err: @@ -1632,6 +1635,9 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); goto err_regmap; } + + return 0; + err_regmap: regmap_exit(wm8753->regmap); err: -- 1.7.4.1

2 5

[alsa-devel] ALC892, Clevo P150HM: Input Source gets reset to "Mic" every resume
by Julian Sikorski 02 Mar '12

02 Mar '12

Hi, whenever I resume the machine, the Input Source would get changed back from "Internal Mic" to "Mic". This is a bit annoying, because pulseaudio is unaware of that change. As a result, everything looks fine in pavucontrol but no sound gets recorded. A quick look at alsamixer -c0 revealed the problem. My alsa info: http://www.alsa-project.org/db/?f=c5776b615a5f3b9ae624e4250fb0c2b69118ce26 This is unrelated to the other problems I have been reporting, since this happens on a mostly-stock Fedora 3.1.2 kernel (the patch I added deals with xhci, not alsa). Regards, Julian

3 8

[alsa-devel] [PATCH v6] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Tomoya MORINAGA 02 Mar '12

02 Mar '12

ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ ML26124-02GD can deliver stable audio performance without being affected by noise from the power supply circuit and peripheral components. The chip also includes a composite video signal output, which can be applied to various portable device requirements. The ML26124 is realized these functions into very small package the size is only 2.56mm x 2.46mm therefore can be construct high quality sound system easily. ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package. Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- V6 - Add break at "case SND_SOC_BIAS_ON:" - Use switch() statement - Delete ASoC cache code. --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ml26124.c | 742 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 132 ++++++++ 4 files changed, 880 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fa787d4..321ea08 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -39,6 +39,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -411,6 +412,9 @@ config SND_SOC_LM4857 config SND_SOC_MAX9877 tristate +config SND_SOC_ML26124 + tristate + config SND_SOC_TPA6130A2 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..b28f48e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -25,6 +25,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -126,6 +127,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 0000000..058b831 --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,742 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ +#define ML26134_CACHESIZE 79 +#define ML26124_VMID BIT(0) +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE +#define ML26124_NUM_REGISTER ML26134_CACHESIZE + +struct ml26124_priv { + u32 mclk; + u32 channels; + struct regmap *regmap; +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); + +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; + +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); + +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limitter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limitter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("Zero Cross Comparator Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limitter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limitter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", + "Analog MIC Differential in", + "Digital MIC in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 2, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLK"}, + {"ADC", NULL, "MCLK"}, + {"DAC", NULL, "PLL"}, + {"ADC", NULL, "PLL"}, + + /* output mixer */ + {"Output Mixer", "PGA Switch", "PGA"}, + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", "Line in Switch", "LIN"}, + + /* outputs */ + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + + /* input */ + {"Input Mux", "Analog MIC SingleEnded in", "MIN"}, + {"Input Mux", "Analog MIC Differential in", "MIN"}, + {"ADC", NULL, "Input Mux"}, +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288800, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +static struct reg_default ml26124_reg[] = { + /* CLOCK control Register */ + {0x00, 0x00 }, /* Sampling Rate */ + {0x02, 0x00}, /* PLL NL */ + {0x04, 0x00}, /* PLLNH */ + {0x06, 0x00}, /* PLLML */ + {0x08, 0x00}, /* MLLMH */ + {0x0a, 0x00}, /* PLLDIV */ + {0x0c, 0x00}, /* Clock Enable */ + {0x0e, 0x00}, /* CLK Input/Output Control */ + + /* System Control Register */ + {0x10, 0x00}, /* Software RESET */ + {0x12, 0x00}, /* Record/Playback Run */ + {0x14, 0x00}, /* Mic Input/Output control */ + + /* Power Management Register */ + {0x20, 0x00}, /* Reference Power Management */ + {0x22, 0x00}, /* Input Power Management */ + {0x24, 0x00}, /* DAC Power Management */ + {0x26, 0x00}, /* SP-AMP Power Management */ + {0x28, 0x00}, /* LINEOUT Power Management */ + {0x2a, 0x00}, /* VIDEO Power Management */ + {0x2e, 0x00}, /* AC-CMP Power Management */ + + /* Analog reference Control Register */ + {0x30, 0x04}, /* MICBIAS Voltage Control */ + + /* Input/Output Amplifier Control Register */ + {0x32, 0x10}, /* MIC Input Volume */ + {0x38, 0x00}, /* Mic Boost Volume */ + {0x3a, 0x33}, /* Speaker AMP Volume */ + {0x48, 0x00}, /* AMP Volume Control Function Enable */ + {0x4a, 0x00}, /* Amplifier Volume Fader Control */ + + /* Analog Path Control Register */ + {0x54, 0x00}, /* Speaker AMP Output Control */ + {0x5a, 0x00}, /* Mic IF Control */ + {0xe8, 0x01}, /* Mic Select Control */ + + /* Audio Interface Control Register */ + {0x60, 0x00}, /* SAI-Trans Control */ + {0x62, 0x00}, /* SAI-Receive Control */ + {0x64, 0x00}, /* SAI Mode select */ + + /* DSP Control Register */ + {0x66, 0x01}, /* Filter Func Enable */ + {0x68, 0x00}, /* Volume Control Func Enable */ + {0x6A, 0x00}, /* Mixer & Volume Control*/ + {0x6C, 0xff}, /* Record Digital Volume */ + {0x70, 0xff}, /* Playback Digital Volume */ + {0x72, 0x10}, /* Digital Boost Volume */ + {0x74, 0xe7}, /* EQ gain Band0 */ + {0x76, 0xe7}, /* EQ gain Band1 */ + {0x78, 0xe7}, /* EQ gain Band2 */ + {0x7A, 0xe7}, /* EQ gain Band3 */ + {0x7C, 0xe7}, /* EQ gain Band4 */ + {0x7E, 0x00}, /* HPF2 CutOff*/ + {0x80, 0x00}, /* EQ Band0 Coef0L */ + {0x82, 0x00}, /* EQ Band0 Coef0H */ + {0x84, 0x00}, /* EQ Band0 Coef0L */ + {0x86, 0x00}, /* EQ Band0 Coef0H */ + {0x88, 0x00}, /* EQ Band1 Coef0L */ + {0x8A, 0x00}, /* EQ Band1 Coef0H */ + {0x8C, 0x00}, /* EQ Band1 Coef0L */ + {0x8E, 0x00}, /* EQ Band1 Coef0H */ + {0x90, 0x00}, /* EQ Band2 Coef0L */ + {0x92, 0x00}, /* EQ Band2 Coef0H */ + {0x94, 0x00}, /* EQ Band2 Coef0L */ + {0x96, 0x00}, /* EQ Band2 Coef0H */ + {0x98, 0x00}, /* EQ Band3 Coef0L */ + {0x9A, 0x00}, /* EQ Band3 Coef0H */ + {0x9C, 0x00}, /* EQ Band3 Coef0L */ + {0x9E, 0x00}, /* EQ Band3 Coef0H */ + {0xA0, 0x00}, /* EQ Band4 Coef0L */ + {0xA2, 0x00}, /* EQ Band4 Coef0H */ + {0xA4, 0x00}, /* EQ Band4 Coef0L */ + {0xA6, 0x00}, /* EQ Band4 Coef0H */ + + /* ALC Control Register */ + {0xb0, 0x00}, /* ALC Mode */ + {0xb2, 0x02}, /* ALC Attack Time */ + {0xb4, 0x03}, /* ALC Decay Time */ + {0xb6, 0x00}, /* ALC Hold Time */ + {0xb8, 0x0b}, /* ALC Target Level */ + {0xba, 0x70}, /* ALC Max/Min Gain */ + {0xbc, 0x00}, /* Noise Gate Threshold */ + {0xbe, 0x00}, /* ALC ZeroCross TimeOut */ + + /* Playback Limiter Control Register */ + {0xc0, 0x04}, /* PL Attack Time */ + {0xc2, 0x05}, /* PL Decay Time */ + {0xc4, 0x0d}, /* PL Target Level */ + {0xc6, 0x70}, /* PL Max/Min Gain */ + {0xc8, 0x10}, /* Playback Boost Volume */ + {0xca, 0x00}, /* PL ZeroCross TimeOut */ + + /* Video Amplifier Control Register */ + {0xd0, 0x01}, /* VIDEO AMP Gain Control */ + {0xd2, 0x01}, /* VIDEO AMP Setup 1 */ + {0xd4, 0x01}, /* VIDEO AMP Control2 */ +}; + +static int ml26124_update_bits(struct snd_soc_codec *codec, unsigned short reg, + unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + int ret; + + ret = snd_soc_read(codec, reg); + if (ret < 0) + return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = snd_soc_write(codec, reg + 1, new); + if (ret < 0) + return ret; + } + + return change; +} + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->channels = params_channels(hw_params); + + switch (params_rate(hw_params)) { + case 16000: + ml26124_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + ml26124_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + ml26124_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + ml26124_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + ml26124_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + ml26124_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + ml26124_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + ml26124_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + ml26124_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + ml26124_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + ml26124_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + ml26124_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + ml26124_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + ml26124_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + ml26124_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + ml26124_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + ml26124_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + ml26124_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + if (mute) + ml26124_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + ml26124_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + + return 0; +} + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned char mask = ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + ml26124_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } +if (!priv) + return -1; + + /* SAI configuration */ + if (priv->channels == 1) { + ml26124_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + ml26124_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + } else { + ml26124_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY); + ml26124_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY); + } + + + return 0; +} + +static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ML26124_USE_PLL: + break; + default: + return -EINVAL; + } + + switch (freq) { + case 12288000: + priv->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +static int ml26124_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (div_id) { + case ML26124_MCLK: + ml26124_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), div); + break; + default: + return -EINVAL; + } + return 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + switch (level) { + case SND_SOC_BIAS_ON: + pr_debug("%s: level=ON priv=%p\n", __func__, priv); + /* VMID ON */ + ml26124_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + break; + case SND_SOC_BIAS_PREPARE: + pr_debug("%s: level=PREPARE\n", __func__); + break; + case SND_SOC_BIAS_STANDBY: + pr_debug("%s: level=STANDBY\n", __func__); + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(priv->regmap); + break; + case SND_SOC_BIAS_OFF: + pr_debug("%s: level=OFF\n", __func__); + /* VMID OFF */ + ml26124_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int ml26124_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ml26124_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 0); + return 0; + break; + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ml26124_update_bits(codec, + ML26124_REC_PLYBAK_RUN, 0x3, 1); + else + ml26124_update_bits(codec, + ML26124_REC_PLYBAK_RUN, 0x3, 2); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, + .set_clkdiv = ml26124_set_dai_clkdiv, + .trigger = ml26124_pcm_trigger, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + regcache_sync(priv->regmap); + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + ml26124_update_bits(codec, ML26124_SW_RST, 0x01, 1); + ml26124_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static const struct regmap_config ml26124_i2c_regmap = { + .val_bits = 8, + .reg_bits = 8, + .max_register = ML26124_NUM_REGISTER, + .reg_defaults = ml26124_reg, + .num_reg_defaults = ARRAY_SIZE(ml26124_reg), + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + struct ml26124_priv *priv = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(priv->regmap); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124-codec", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +static int __init ml26124_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&ml26124_i2c_driver); + if (ret != 0) + pr_err("Failed to register ML26124 I2C driver: %d\n", ret); + + return ret; +} +module_init(ml26124_modinit); + +static void __exit ml26124_exit(void) +{ + i2c_del_driver(&ml26124_i2c_driver); +} +module_exit(ml26124_exit); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 0000000..d4a8df5 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,132 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +/* Clock select for machine driver */ +#define ML26124_USE_PLL 0 +#define ML26124_USE_MCLKI_256FS 1 +#define ML26124_USE_MCLKI_512FS 2 +#define ML26124_USE_MCLKI_1024FS 3 + +enum ml26124_regs { + ML26124_MCLK = 0, +}; + +#endif -- 1.7.7.6

2 5

[alsa-devel] [PATCH v2 00/10] OMAP/ASoC: Move and merge McBSP driver under ASoC
by Peter Ujfalusi 02 Mar '12

02 Mar '12

Hello, Changes since v1 (RFC series): - removed pm_runtime_get/put_sync calls since core takes care of this. Fixes the initial issue with OMAP3 Pandora setup reported by Grazvydas Ignotas - Dropped my version of snd_soc_add_dai_controls from the series since Liam implementation adds the same functionality. - Added Tested-by tags from Janusz (OMAP1), and Grazvydas (OMAP3 pandora) - Added Ack from Tony for the arch/arm/ patches - I have not added Ack from Mark, and Jarkko although they said it looks OK, but not given explicit ack for the series. NOTE: this series has been generated agains Takashi's topic/asoc branch merged with Mark's for-next branch since this series depends on changes in Marks' branch, but not yet pulled by Takashi (snd_soc_add_dai_controls implementation from Liam). Intro mail from thre RFC series: The aim of this series is to move the plat-omap/mcbsp.c driver out to sound/soc/omap/ By doing this we can combine the two levels of McBSP driver stack into a single one. Formally we had omap-mcbsp.[1-X] drivers (plat-omap/mcbsp.c), and omap-mcbsp-dai.[0-(X-1) drivers (sound/soc/omap/omap-mcbsp.c). We had global, shared structures, variables, pointers. The API between the two driver was a set of exported function calls where one of the parameter was the McBSP port id. With this series we will have one driver for McBSP (omap-mcbsp.[1-X]) which is basically the ASoC dai driver. All exported/shared global structures are gone since we can use driver data for all ports in this way. With the code in one place we can do further clean up in the McBSP driver, and adding new features will be much easier in the future. Also the DT support for McBSP will be significantly easier if we have one set of driver. McBSP is only used by audio, there are no known other users of the OMAP McBSP (well, Nokia n700 has the WLAN connected through McBSP but we do not have driver for that anyway). The series was created against Takashi's topic/asoc branch. I have tested it on Beagle xM, and compile tested it for OMAP1, OMAP2, and OMAP4 as well. I'm going to do more work on the cleanup, and the plan is to merge the two files into one (mcbsp.c -> omap-mcbsp.c) eventually. Regards, Peter --- Peter Ujfalusi (10): ARM: OMAP: mcbsp: Convert core driver to proper platform driver OMAP: mcbsp: Move core driver under sound/soc/omap ASoC: OMAP: McBSP: Consolidate plat/mcbsp.h content ASoC: omap-mcbsp: Create a single driver for McBSP ARM: OMAP: Do not register omap-mcbsp-dai device ASoC: OMAP: mcbsp.h: Clean up bit definitions ASoC: OMAP McBSP: Remove redundant accessors ASoC: omap-mcbsp: Cleanup of module probe/remove code ASoC: omap-mcbsp: Merge the omap_mcbsp_data into omap_mcbsp structure ASoC: omap-mcbsp: Simplify DMA configuration arch/arm/mach-omap1/devices.c | 9 - arch/arm/mach-omap1/mcbsp.c | 14 +- arch/arm/mach-omap2/devices.c | 22 -- arch/arm/mach-omap2/mcbsp.c | 8 +- arch/arm/plat-omap/Kconfig | 8 - arch/arm/plat-omap/Makefile | 2 - arch/arm/plat-omap/include/plat/mcbsp.h | 321 ---------------- sound/soc/omap/Kconfig | 4 + sound/soc/omap/Makefile | 1 + sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/igep0020.c | 2 +- {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c | 470 ++++-------------------- sound/soc/omap/mcbsp.h | 337 +++++++++++++++++ sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-mcbsp.c | 287 ++++++++------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.h | 2 + sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/rx51.c | 4 +- sound/soc/omap/sdp3430.c | 4 +- sound/soc/omap/zoom2.c | 4 +- 26 files changed, 598 insertions(+), 921 deletions(-) rename {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c (68%) create mode 100644 sound/soc/omap/mcbsp.h -- 1.7.8.4

3 19

[alsa-devel] [PATCH] ALSA: hda - Add a fake mute feature
by Takashi Iwai 01 Mar '12

01 Mar '12

David, I thought you have a machine with Cx20561 (such as Lenovo X200, T400 or W500)? Could you check whether this really doesn't break? Thanks, Takashi === Some codecs don't supply the mute amp-capabilities although the lowest volume gives the mute. It'd be handy if the parser provides the mute mixers in such a case. This patch adds an extension amp-cap bit (which is used only in the driver) to represent the min volume = mute state. Also modified the amp cache code to support the fake mute feature when this bit is set but the real mute bit is unset. In addition, conexant cx5051 parser uses this new feature to implement the missing mute controls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42825 Cc: <stable(a)kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 8 ++++++-- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/patch_conexant.c | 22 +++++++++++++++++++++- 3 files changed, 30 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c2c65f6..0ae6eb2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; parm |= index << AC_AMP_SET_INDEX_SHIFT; - parm |= val; + if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) && + (info->amp_caps & AC_AMPCAP_MIN_MUTE)) + ; /* set the zero value as a fake mute */ + else + parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); info->vol[ch] = val; } @@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); val1 += ofs; val1 = ((int)val1) * ((int)val2); - if (min_mute) + if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) val2 |= TLV_DB_SCALE_MUTE; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e9f71dc..f0f1943 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -298,6 +298,9 @@ enum { #define AC_AMPCAP_MUTE (1<<31) /* mute capable */ #define AC_AMPCAP_MUTE_SHIFT 31 +/* driver-specific amp-caps: using bits 24-30 */ +#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ + /* Connection list */ #define AC_CLIST_LENGTH (0x7f<<0) #define AC_CLIST_LONG (1<<7) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a7a5733..ca117cf 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; - if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE)) + if (!(query_amp_caps(codec, nid, hda_dir) & + (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))) break; } return 0; @@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = { {} }; +/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches + * can be created (bko#42825) + */ +static void add_cx5051_fake_mutes(struct hda_codec *codec) +{ + static hda_nid_t out_nids[] = { + 0x10, 0x11, 0 + }; + hda_nid_t *p; + + for (p = out_nids; *p; p++) + snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT, + AC_AMPCAP_MIN_MUTE | + query_amp_caps(codec, *p, HDA_OUTPUT)); +} + static int patch_conexant_auto(struct hda_codec *codec) { struct conexant_spec *spec; @@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15045: spec->single_adc_amp = 1; break; + case 0x14f15051: + add_cx5051_fake_mutes(codec); + break; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); -- 1.7.9

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[alsa-devel] [PATCH] ASoC: Samsung: Update email id of the author
by Jaswinder Singh 01 Mar '12

01 Mar '12

I moved on from a great employer and the email-id no longer exists. Update email-id to a personal one, assuming I don't move on from myself anytime soon. And when I do, people don't get the eulogies bounced. Signed-off-by: Jaswinder Singh <jassisinghbrar(a)gmail.com> --- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/samsung/i2s.c | 4 ++-- sound/soc/samsung/i2s.h | 2 +- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/smdk_wm8580.c | 4 ++-- sound/soc/samsung/smdk_wm9713.c | 4 ++-- 6 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 7b9bf93..e8883d4d 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -4,7 +4,7 @@ * Evolved from s3c2443-ac97.c * * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Author: Jaswinder Singh <jassisinghbrar(a)gmail.com> * Credits: Graeme Gregory, Sean Choi * * This program is free software; you can redistribute it and/or modify @@ -511,7 +511,7 @@ static struct platform_driver s3c_ac97_driver = { module_platform_driver(s3c_ac97_driver); -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:samsung-ac97"); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 87a874d..7ccb075 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. - * Jaswinder Singh <jassi.brar(a)samsung.com> + * Jaswinder Singh <jassisinghbrar(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -1143,7 +1143,7 @@ static struct platform_driver samsung_i2s_driver = { module_platform_driver(samsung_i2s_driver); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); MODULE_DESCRIPTION("Samsung I2S Interface"); MODULE_ALIAS("platform:samsung-i2s"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/i2s.h b/sound/soc/samsung/i2s.h index 8e15f6a..d420a7c 100644 --- a/sound/soc/samsung/i2s.h +++ b/sound/soc/samsung/i2s.h @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. - * Jaswinder Singh <jassi.brar(a)samsung.com> + * Jaswinder Singh <jassisinghbrar(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 5678020..b7b2a1f 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - S3C PCM-Controller driver * * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Author: Jaswinder Singh <jassisinghbrar(a)gmail.com> * based upon I2S drivers by Ben Dooks. * * This program is free software; you can redistribute it and/or modify @@ -639,7 +639,7 @@ static struct platform_driver s3c_pcm_driver = { module_platform_driver(s3c_pcm_driver); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); MODULE_DESCRIPTION("S3C PCM Controller Driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:samsung-pcm"); diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index bff8758..ade2809 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -2,7 +2,7 @@ * smdk_wm8580.c * * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Author: Jaswinder Singh <jassisinghbrar(a)gmail.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -253,6 +253,6 @@ static void __exit smdk_audio_exit(void) } module_exit(smdk_audio_exit); -MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_AUTHOR("Jaswinder Singh, jassisinghbrar(a)gmail.com"); MODULE_DESCRIPTION("ALSA SoC SMDK WM8580"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c index 8e26a73..55b2ca7 100644 --- a/sound/soc/samsung/smdk_wm9713.c +++ b/sound/soc/samsung/smdk_wm9713.c @@ -2,7 +2,7 @@ * smdk_wm9713.c -- SoC audio for SMDK * * Copyright 2010 Samsung Electronics Co. Ltd. - * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.com> + * Author: Jaswinder Singh Brar <jassisinghbrar(a)gmail.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License as @@ -103,6 +103,6 @@ module_init(smdk_init); module_exit(smdk_exit); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); +MODULE_AUTHOR("Jaswinder Singh Brar, jassisinghbrar(a)gmail.com"); MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); MODULE_LICENSE("GPL"); -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Samsung: Merge two identical if-else clauses
by Jassi Brar 01 Mar '12

01 Mar '12

Saves two lines and a hell of a lot of embarrassment looking at the code. Signed-off-by: Jassi Brar <jaswinder.singh(a)linaro.org> --- sound/soc/samsung/i2s.c | 10 ++++------ 1 files changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 7ccb075..6553b19 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -761,15 +761,13 @@ static int i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: local_irq_save(flags); - if (capture) + if (capture) { i2s_rxctrl(i2s, 0); - else - i2s_txctrl(i2s, 0); - - if (capture) i2s_fifo(i2s, FIC_RXFLUSH); - else + } else { + i2s_txctrl(i2s, 0); i2s_fifo(i2s, FIC_TXFLUSH); + } local_irq_restore(flags); break; -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: wm8996: Fix /RESET bounce ordering
by Mark Brown 29 Feb '12

29 Feb '12

We want to leave the device out of rather than in reset. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c1c7ef0..9376b19 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1719,8 +1719,8 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) static int wm8996_reset(struct wm8996_priv *wm8996) { if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); return 0; } else { return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, -- 1.7.9.1

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[alsa-devel] [PATCH] ASoC: wm_hubs: Bomb out if we can't read back the DC servo result
by Mark Brown 29 Feb '12

29 Feb '12

Should have no practical impact but it's safer than trying to soldier on. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 689311c..9d60a6e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -172,7 +172,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) break; default: WARN(1, "Unknown DCS readback method\n"); - break; + return; } dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); -- 1.7.9.1

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[alsa-devel] [PATCH] ASoC: wm8994: Make sure we don't have MICBIAS on during jackdet mode
by Mark Brown 29 Feb '12

29 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++------------ 1 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 33bc718..691e897 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3170,9 +3170,6 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); - wm1811_jackdet_set_mode(codec, - WM1811_JACKDET_MODE_JACK); - if (wm8994->pdata->jd_ext_cap) { mutex_lock(&codec->mutex); snd_soc_dapm_disable_pin(&codec->dapm, @@ -3180,6 +3177,9 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_dapm_sync(&codec->dapm); mutex_unlock(&codec->mutex); } + + wm1811_jackdet_set_mode(codec, + WM1811_JACKDET_MODE_JACK); } } @@ -3235,23 +3235,24 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); - /* If required for an external cap force MICBIAS on */ - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } - /* * Start off measument of microphone impedence to find * out what's actually there. */ wm8994->mic_detecting = true; wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_MIC); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); + + /* If required for an external cap force MICBIAS on */ + if (wm8994->pdata->jd_ext_cap) { + mutex_lock(&codec->mutex); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS2"); + snd_soc_dapm_sync(&codec->dapm); + mutex_unlock(&codec->mutex); + } } else { dev_dbg(codec->dev, "Jack not detected\n"); -- 1.7.9.1

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