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

[alsa-devel] [PATCH v4] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Tomoya MORINAGA 30 Nov '11

30 Nov '11

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> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ml26124.c | 595 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 122 +++++++++ 4 files changed, 723 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 665d924..ce8e622 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -38,6 +38,7 @@ 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_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -211,6 +212,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_ML26124 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e..c1353ff 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ 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-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -122,6 +123,7 @@ 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-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 0000000..7753a9c --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,595 @@ +/* + * 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 <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +struct ml26124_priv { + u32 mclk; + u32 channels; +}; + +/* 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 Input Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Input Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Input Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Input 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", 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 in", + "Digital MIC in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 1, 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_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 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"), + SND_SOC_DAPM_OUTPUT("MICBIASO"), +}; + +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", NULL, "Line in"}, + + /* outputs */ + {"SPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + {"MICBIASO", NULL, "MICBIAS"}, + {"LOUT", NULL, "Line Out Enable"}, + + /* input mux */ + {"Input Mux", "Analog MIC in", "MICBIAS"}, + {"Input Mux", "Digital MIC in", "MICBIAS"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"DAC", NULL, "MDIN"}, + {"PGA", NULL, "MIN"}, + {"SPOUT", "Line Out Enable", "LIN"}, +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* 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}, +}; + +/* 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: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_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; +} + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + return 0; +} + +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ + +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; + } + snd_soc_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; + } + + /* SAI configuration */ + if (priv->channels == 1) { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + } else { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY); + snd_soc_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 (freq) { + case 12288000: + priv->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +#define ML26134_CACHESIZE 79 +static const u16 ml26124_reg[ML26134_CACHESIZE] = { + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, + 0x0, 0x0, 0x0, /* System Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* Power Management */ + 0x4, /* Analog Reference Control */ + 0x10, 0x0, 0x33, 0x0, 0x0, /* Input/Output Amplifier Control */ + 0x0, 0x0, 0x1, /* Analog Path Contorl */ + 0x0, 0x0, 0x0, /* Audio Interface Control */ + 0x1, 0x0, 0x0, 0xff, 0xff, 0x10, /* DSP Control */ + 0x7, 0x7, 0x7, 0x7, 0x7, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x2, 0x3, 0x0, 0xb, 0x70, 0x0, 0x0, /* ALC Control */ + 0x4, 0x5, 0xd, 0x70, 0x10, 0x0, /* Playback Limiter Control */ + 0x1, 0x1, 0x1 /* Video Amplifer Control */ +}; + +#define ML26124_VMID BIT(0) + +static void ml26124_sync_cache(struct snd_soc_codec *codec) +{ + u8 *reg_cache = codec->reg_cache; + int i; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + + for (i = 1; i < codec->driver->reg_cache_size; i++) { + if (reg_cache[i] == ml26124_reg[i]) + continue; + + snd_soc_write(codec, i, reg_cache[i]); + } + + codec->cache_sync = 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* VMID ON */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ml26124_sync_cache(codec); + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_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; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0); + return 0; + } else if (cmd == SNDRV_PCM_TRIGGER_START) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 1); + else + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 2); + } + + return 0; +} + +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +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, + .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) +{ + 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; + + 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 */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_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, + .reg_cache_size = ML26134_CACHESIZE, + .reg_word_size = sizeof(u8), + .reg_cache_default = ml26124_reg, + .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 __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + 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", + .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 = 0; + + 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-linux(a)dsn.lapis-semi.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..d30b530 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,122 @@ +/* + * 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 + +#endif -- 1.7.4.4

2 2

[alsa-devel] [PATCH 1/7] ASoC: ad193X: Use table based DAPM and controls setup
by Lars-Peter Clausen 30 Nov '11

30 Nov '11

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 1901cd2..1dfda5c 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -345,7 +345,6 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; codec->control_data = ad193x->regmap; @@ -371,17 +370,17 @@ static int ad193x_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04); - snd_soc_add_controls(codec, ad193x_snd_controls, - ARRAY_SIZE(ad193x_snd_controls)); - snd_soc_dapm_new_controls(dapm, ad193x_dapm_widgets, - ARRAY_SIZE(ad193x_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); - return ret; } static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .probe = ad193x_probe, + .controls = ad193x_snd_controls, + .num_controls = ARRAY_SIZE(ad193x_snd_controls), + .dapm_widgets = ad193x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad193x_dapm_widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), }; #if defined(CONFIG_SPI_MASTER) -- 1.7.7.3

3 13

Re: [alsa-devel] Kernel: snd-usb-audio: probe of 2-2:1.0 failed with error -5
by Linus Torvalds 30 Nov '11

30 Nov '11

On Tue, Nov 29, 2011 at 4:52 AM, Nixbits Admin <admin(a)nixbits.org> wrote: > > While knowing that your time is precious and without wanting to making you > spend more than what you may be able to spare for my problem; i would like > ask you if you could take a look at this issue. > > https://forums.gentoo.org/viewtopic-p-6885908.html#6885908 > > snd-usb-audio: probe of 2-2:1.0 failed with error -5 This is related to the line above it: Nov 29 07:21:21 kernel: skipping empty audio interface (v1) Nov 29 07:21:21 kernel: snd-usb-audio: probe of 2-3:1.0 failed with error -5 which in turn is because of a zero "bInCollection" value in the device control header. I'm adding some sound people to the cc, in case anybody has more clues about what's wrong with that particular device - I don't personally have any real clue about usb audio. As to your UHCI/EHCI confusion - it's actually a USB confusion. EHCI was done as a add-on to UHCI (or OHCI), to the point where there is both a UHCI *and* an EHCI controller involved (on the same silicon) and they have this odd hand-over thing where one part (the UHCI part) handles legacy devices, and then the new bolted-on EHCI part handles the full USB-2.0 functionality. So the UHCI/EHCI confusion is deep in the silicon itself. If you attach a usb-1.1 device, it will be handed off to the UHCI part, if you attach a 2.0 device it will be handled by the EHCI part of the controller. Oh the wonders of crazy backwards compatibility hacks.. Linus

4 5

[alsa-devel] [PATCH] ASoC: Fix __iomem annotation for IDMA registers
by Mark Brown 30 Nov '11

30 Nov '11

We always store the register address as __iomem but pass it around as a plain void * which upsets sparse. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/idma.c | 2 +- sound/soc/samsung/idma.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index fb80f28..5de500c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -881,7 +881,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) writel(CON_RSTCLR, i2s->addr + I2SCON); if (i2s->quirks & QUIRK_SEC_DAI) - idma_reg_addr_init((void *)i2s->addr, + idma_reg_addr_init(i2s->addr, i2s->sec_dai->idma_playback.dma_addr); probe_exit: diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 6ca3d8c..baf97eb 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -403,7 +403,7 @@ static int idma_new(struct snd_soc_pcm_runtime *rtd) return ret; } -void idma_reg_addr_init(void *regs, dma_addr_t addr) +void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) { spin_lock_init(&idma.lock); idma.regs = regs; diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h index 4827321..8644946 100644 --- a/sound/soc/samsung/idma.h +++ b/sound/soc/samsung/idma.h @@ -14,7 +14,7 @@ #ifndef __SND_SOC_SAMSUNG_IDMA_H_ #define __SND_SOC_SAMSUNG_IDMA_H_ -extern void idma_reg_addr_init(void *regs, dma_addr_t addr); +extern void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr); /* dma_state */ #define LPAM_DMA_STOP 0 -- 1.7.7.3

2 1

[alsa-devel] Default device XRUNs with perriods=2
by Alan Horstmann 29 Nov '11

29 Nov '11

Hi, I am trying to debug/fix some problems with Portaudio on Linux platform, and would appreciate assistance from Alsa experts. The 'default' pcm on the test systems here (Suse 9 and SLED10), is standard for the configuration and includes dmix and plug conversion, with fixed period of 1024 at 48000. Pulseaudio is not on these systems. When playing data (using a polling mechanism) at 48000 both 3 (or more) and 2 periods of buffer work fine; with the fixed period of 1024 the lowest latency is 2 * 1024 / 48000 = 42.6ms. However, at 44100, 2 periods does not run properly at all, but 3 does. The periods in this case are (940, 941) due to the rate conversion. What I see (captured on another machine) from start of the stream is one 21.3ms period of good audio, a discontinuity followed by some variable amount of out-of-sync audio, and a section of silence. I believe that the discontinuity corresponds to the point at which snd_pcm_poll_descriptors_revents() returns POLLERR and portaudio detects the POLLERR and restarts the stream, so that after a certain delay there is another 21.3ms of audio, POLLERR, now no out-of-sync audio, just silence, before it all repeats. So it stutters only outputting 21.3ms bursts of audio. The CPU usage during this is about 2%. I am suspecting some mis-configuration, or is it wrong to expect this to work at 2 periods due to the rate conversion etc? Any pointers etc would be really appreciated. This is the info dump:- ACCESS: MMAP_INTERLEAVED FORMAT: FLOAT_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: 64 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (21333 21334) PERIOD_SIZE: (940 941) PERIOD_BYTES: (7520 7528) PERIODS: 2 BUFFER_TIME: (42653 42654) BUFFER_SIZE: 1881 BUFFER_BYTES: 15048 TICK_TIME: 0 start_mode: DATA xrun_mode: STOP tstamp_mode: MMAP period_step: 1 sleep_min: 0 avail_min: 940 xfer_align: 1 silence_threshold: 0 silence_size: 4235635449541951488 boundary: 4235635449541951488 stream : PLAYBACK access : MMAP_INTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 1881 period_size : 940 period_time : 21333 tick_time : 0 tstamp_mode : MMAP period_step : 1 sleep_min : 0 avail_min : 940 xfer_align : 1 start_threshold : 940 stop_threshold : 1881 silence_threshold: 0 silence_size : 4235635449541951488 boundary : 4235635449541951488 Regards Alan

3 8

Re: [alsa-devel] ASoC: Dynamic CODEC DAI
by Patrick Lai 29 Nov '11

29 Nov '11

On 11/28/2011 5:19 PM, Pierre-Louis Bossart wrote: >> The CODEC which I am working on transports digital audio through >> SLIMBUS instead of I2S. The concept of CODEC DAI does not apply too >> well with SLIMBUS architecture as CODEC DAIs are typically defined base >> on a set of I2S digital audio interface(bit clock, world select, sd >> lines). On the CODEC, there are 10 digital capture ports. They can be >> independently configured as mono channels or can be run-time grouped >> together to function like multi-channel CODEC DAIs. On top of that, >> only some ports out of 10 ports can accept all analog mic inputs or >> digital mic inputs. Hence, even though the CODEC satisfies the >> concurrent use cases it was designed for, software would have to be >> articulate on grouping the ports in order to utilize all ports. So, I >> cannot simply code up CODEC DAI definitions in the CODEC driver for a >> particular machine. For now, I am looking for compile time grouping of >> these ports as use cases are known at the time machine is designed. > > I may be reaching summits of cluelessness here, but isn't the logical > grouping/shuffling of channels needed on the cpu/host side only, possibly at > the machine-driver level? This is exactly what I am planning to do. Have the machine driver handles the grouping. However, DAI link is consisted of one CODEC dai, one CPU DAI, and platform driver. Unlike I2s which number of channels per DAI is fixed, I can divide 10 available ports to several groups. Each group belongs to different DAI links so ports within same group form x number of channels stream and feed into CPU. However, how many ports per group and which ports to use is unknown to CODEC driver and would have to be decided by machine driver. . That's why I need to propose the design on ASoc framework. At the codec level you should only need to worry > about which channel/port goes to what output, and that would be similar to > TDM/AC97. I don't understand why all this should be part of a codec driver. Please excuse my lack of understanding about AC97. There is a codec driver /sound/soc/codec/ac97.c. However, only one DAI is defined and supports both playback/capture with 2 as max channels. I don't see how static CODEC DAI definition is going to satisfy my requirements. I am still bounded to DAI link structure definition which takes only one CODEC DAI. In this case with ac97.c, I am still limited to work with pre-defined CODEC DAI in ac97.c Is there an example for grouping ac97 channels? > -Pierre > > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 1

Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Takashi Iwai 29 Nov '11

29 Nov '11

At Mon, 28 Nov 2011 15:36:15 -0600, Pierre-Louis Bossart wrote: > > > > implementation? At least, the term "frame" is already used in ALSA > > > PCM, and I'm not sure whether you use this term for the very same > > > meaning in the above context... > > Most compressed formats have a notion of frame, but this is indeed a > different notion. In ALSA a frame is really a sampling point, possibly with > multiple channels. Compression algorithms group sampling points in frames, > or blocks, before applying a transform and quantizing. AAC works with 1024 > sampling points, MPEG Layer3 with 2 granules of 576 points, AMR with 160 > points, etc. > We may want to use terms like 'blocks' or 'chunks', if this helps avoid > confusions with existing ALSA concepts. Yes, it's better to avoid the conflicting definition, IMO. > > For capture, since application may need to get data on frame basis > > (think video recording with encoded video usage, where application > > needs > > audio compressed data on "frame" basis for encapsulation). The DSP is > > supposed callback after every encoded frame. I think only difference > > between PCM and this is encoding format, otherwise in terms of decoded > > data and time they would mean the same.. I am not expert here so maybe > > wrong. > > For capture, in some cases the compressed bitstream doesn't provide any > pointers on the beginning of a block, nor any block-length indication (eg. > AAC-RAW). In that case, an encoder would need to pass data to user-space on > a block-by-block basis, with the bytes available in the ring buffer > corresponding to the block length. If the applications can extract 'blocks' > on their own and handle the relevant file-write/multiplexing, the usual > data-passing with regular events is fine. > > For playback, the decoder is expected to deal with such situations on its > own and find the block boundaries, meaning at the application level we can > just push bytes down to the decoder without worrying. But is this restriction guaranteed to be applicable to all possible hardwares in future? What happens if you'll get a hardware that doesn't support the byte-unit push for the playback? That said, I see no obvious reason to give a restriction coupled with the stream direction. thanks, Takashi

2 1

[alsa-devel] [PATCH] ASoC: Enhance default WM8958 microphone detection
by Mark Brown 29 Nov '11

29 Nov '11

Actively manage the detection rate for microphones with WM8958, providing improved power consumption and maximising the benefit from the hardware debounce. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 120 ++++++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/wm8994.h | 2 + 2 files changed, 111 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 207bccd..027bf68 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -53,6 +53,56 @@ static int wm8994_retune_mobile_base[] = { WM8994_AIF2_EQ_GAINS_1, }; +static void wm8958_default_micdet(u16 status, void *data); + +static const struct { + int sysclk; + bool idle; + int start; + int rate; +} wm8958_micd_rates[] = { + { 32768, true, 1, 4 }, + { 32768, false, 1, 1 }, + { 44100 * 256, true, 7, 6 }, + { 44100 * 256, false, 7, 6 }, +}; + +static void wm8958_micd_set_rate(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int best, i, sysclk, val; + bool idle; + + if (wm8994->jack_cb != wm8958_default_micdet) + return; + + idle = !wm8994->jack_mic; + + sysclk = snd_soc_read(codec, WM8994_CLOCKING_1); + if (sysclk & WM8994_SYSCLK_SRC) + sysclk = wm8994->aifclk[1]; + else + sysclk = wm8994->aifclk[0]; + + best = 0; + for (i = 0; i < ARRAY_SIZE(wm8958_micd_rates); i++) { + if (wm8958_micd_rates[i].idle != idle) + continue; + if (abs(wm8958_micd_rates[i].sysclk - sysclk) < + abs(wm8958_micd_rates[best].sysclk - sysclk)) + best = i; + else if (wm8958_micd_rates[best].idle != idle) + best = i; + } + + val = wm8958_micd_rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT + | wm8958_micd_rates[best].rate << WM8958_MICD_RATE_SHIFT; + + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_BIAS_STARTTIME_MASK | + WM8958_MICD_RATE_MASK, val); +} + static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -221,8 +271,10 @@ static int configure_clock(struct snd_soc_codec *codec) */ /* If they're equal it doesn't matter which is used */ - if (wm8994->aifclk[0] == wm8994->aifclk[1]) + if (wm8994->aifclk[0] == wm8994->aifclk[1]) { + wm8958_micd_set_rate(codec); return 0; + } if (wm8994->aifclk[0] < wm8994->aifclk[1]) new = WM8994_SYSCLK_SRC; @@ -236,6 +288,8 @@ static int configure_clock(struct snd_soc_codec *codec) snd_soc_dapm_sync(&codec->dapm); + wm8958_micd_set_rate(codec); + return 0; } @@ -2987,21 +3041,56 @@ static void wm8958_default_micdet(u16 status, void *data) { struct snd_soc_codec *codec = data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int report = 0; /* If nothing present then clear our statuses */ - if (!(status & WM8958_MICD_STS)) - goto done; + if (!(status & WM8958_MICD_STS)) { + dev_dbg(codec->dev, "Detected open circuit\n"); + wm8994->jack_mic = false; + wm8994->detecting = true; + + wm8958_micd_set_rate(codec); - report = SND_JACK_MICROPHONE; + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_BTN_0 | SND_JACK_HEADSET); + + return; + } - /* Everything else is buttons; just assign slots */ - if (status & 0x1c) - report |= SND_JACK_BTN_0; + /* If the measurement is showing a high impedence we've got a + * microphone. + */ + if (wm8994->detecting && (status & 0x600)) { + dev_dbg(codec->dev, "Detected microphone\n"); + + wm8994->detecting = false; + wm8994->jack_mic = true; + + wm8958_micd_set_rate(codec); + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADSET, + SND_JACK_HEADSET); + } -done: - snd_soc_jack_report(wm8994->micdet[0].jack, report, - SND_JACK_BTN_0 | SND_JACK_MICROPHONE); + + if (wm8994->detecting && status & 0x4) { + dev_dbg(codec->dev, "Detected headphone\n"); + wm8994->detecting = false; + + wm8958_micd_set_rate(codec); + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); + } + + /* Report short circuit as a button */ + if (wm8994->jack_mic) { + if (status & 0x4) + snd_soc_jack_report(wm8994->micdet[0].jack, + SND_JACK_BTN_0, SND_JACK_BTN_0); + else + snd_soc_jack_report(wm8994->micdet[0].jack, + 0, SND_JACK_BTN_0); + } } /** @@ -3047,6 +3136,15 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8994->jack_cb = cb; wm8994->jack_cb_data = cb_data; + wm8994->detecting = true; + wm8994->jack_mic = false; + + wm8958_micd_set_rate(codec); + + /* Detect microphones and short circuits */ + snd_soc_update_bits(codec, WM8958_MIC_DETECT_2, + WM8958_MICD_LVL_SEL_MASK, 0x41); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); } else { diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f4f1355..1087425 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -126,6 +126,8 @@ struct wm8994_priv { struct soc_enum enh_eq_enum; struct wm8994_micdet micdet[2]; + bool detecting; + bool jack_mic; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- 1.7.7.3

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[PATCH] ALSA: HDA/Jack: Make device indices for HDMI
by David Henningsson 29 Nov '11

29 Nov '11

Make sure we can associate HDMI Jacks with the correct PCM devices. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 5 +++-- sound/pci/hda/hda_jack.h | 2 +- sound/pci/hda/patch_hdmi.c | 3 ++- 3 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3949015..0b1f72c 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -210,7 +210,7 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); * will have the given name and index. */ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx) + const char *name, int idx, int device) { struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; @@ -223,6 +223,7 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, kctl = snd_kctl_jack_new(name, idx, codec); if (!kctl) return -ENOMEM; + kctl->id.device = device; if (snd_hda_ctl_add(codec, nid, kctl) < 0) return -ENOMEM; jack->kctl = kctl; @@ -249,7 +250,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, return 0; snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); - err = snd_hda_jack_add_kctl(codec, nid, name, idx); + err = snd_hda_jack_add_kctl(codec, nid, name, idx, 0); if (err < 0) return err; return snd_hda_jack_detect_enable(codec, nid, 0); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index f8f97c7..fcca96c 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -76,7 +76,7 @@ static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) } int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx); + const char *name, int idx, int device); int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bb8cfc6..9689c8d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1277,7 +1277,8 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; err = snd_hda_jack_add_kctl(codec, per_pin->pin_nid, - "HDMI", pin_idx); + "HDMI", pin_idx, + spec->pcm_rec[pin_idx].device); if (err < 0) return err; } -- 1.7.5.4 --------------050907080508000907070701--

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[PATCH] alsa-driver-build: Make new jack API compile on older kernels
by David Henningsson 29 Nov '11

29 Nov '11

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- acore/ctljack.c | 4 ++++ pci/hda/hda_jack.c | 3 +++ 2 files changed, 7 insertions(+), 0 deletions(-) create mode 100644 acore/ctljack.c create mode 100644 pci/hda/hda_jack.c diff --git a/acore/ctljack.c b/acore/ctljack.c new file mode 100644 index 0000000..4d9352f --- /dev/null +++ b/acore/ctljack.c @@ -0,0 +1,4 @@ +#define __NO_VERSION__ +#include "adriver.h" +#include "../alsa-kernel/core/ctljack.c" + diff --git a/pci/hda/hda_jack.c b/pci/hda/hda_jack.c new file mode 100644 index 0000000..160ad73 --- /dev/null +++ b/pci/hda/hda_jack.c @@ -0,0 +1,3 @@ +#include "adriver.h" +#include "../../alsa-kernel/pci/hda/hda_jack.c" + -- 1.7.5.4 --------------050907080508000907070701 Content-Type: text/x-patch; name="0001-ALSA-HDA-Jack-Make-device-indices-for-HDMI.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0001-ALSA-HDA-Jack-Make-device-indices-for-HDMI.patch"

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