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

[alsa-devel] [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Tomoya MORINAGA 02 Dec '11

02 Dec '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 (PSRR) 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 | 475 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 122 +++++++++++ 4 files changed, 603 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..ee323fb --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,475 @@ +/* + * 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 { + struct snd_pcm_substream *substream; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_vol, -7150, 50, 0); +static const DECLARE_TLV_DB_SCALE(eq_band_gain, -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_vol), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_vol), + 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, eq_band_gain), + SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, eq_band_gain), + SOC_SINGLE_TLV("EQ Band2 Input Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, eq_band_gain), + SOC_SINGLE_TLV("EQ Band3 Input Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, eq_band_gain), + SOC_SINGLE_TLV("EQ Band4 Input Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, eq_band_gain), + 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("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 Switch", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limitter Switch", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Volume 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 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_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 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("VIDEOIN"), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("VIDEOOUT"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* output mixer */ + {"Output Mixer", "PGA Switch", "PGA"}, + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", NULL, "Line in Switch"}, + /* outputs */ + {"SPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + + /* 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", NULL, "LIN"}, +}; + +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; + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 3); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 6); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 0x08); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0x00); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x30); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + /* Record/Playback Running Control Register */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x2, 0x2); + else + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x1, 0x1); + + return 0; +} + +static int snd_card_ml7213i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* soft reset assert */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + + /* Stop Record/Playback Running */ + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0); + + 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)) + + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned char mode; + 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 */ + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, mask); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, mask); + + 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 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); + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + 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; +} + +#define ML26124_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + 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, + .hw_free = snd_card_ml7213i2s_hw_free, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, +}; + +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); + + snd_soc_add_controls(codec, ml26124_snd_controls, + ARRAY_SIZE(ml26124_snd_controls)); + + return 0; +} + +/* power down chip */ +#define REG_CACHE_SIZE 0x79 +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 = REG_CACHE_SIZE, + .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), +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + int ret; + + priv = kzalloc(sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); + if (ret < 0) + kfree(priv); + + return ret; +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + 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

4 17

[alsa-devel] [PATCH 1/3] ASoC: Add signal generator widget type
by Mark Brown 01 Dec '11

01 Dec '11

A signal generator behaves as an input would but is not considered for any of the special behaviour associated with external input pins. This is especially useful when automatically working out not connected widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 4 ++++ sound/soc/soc-dapm.c | 7 +++++++ 2 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0c159a7..d26a9b7 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -43,6 +43,9 @@ .num_kcontrols = 0} /* platform domain */ +#define SND_SOC_DAPM_SIGGEN(wname) \ +{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \ + .num_kcontrols = 0, .reg = SND_SOC_NOPM } #define SND_SOC_DAPM_INPUT(wname) \ { .id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \ .num_kcontrols = 0, .reg = SND_SOC_NOPM } @@ -410,6 +413,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_supply, /* power/clock supply */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ + snd_soc_dapm_siggen, /* signal generator */ }; /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index da5c1ae..6bb327e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -339,6 +339,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_output: case snd_soc_dapm_adc: case snd_soc_dapm_input: + case snd_soc_dapm_siggen: case snd_soc_dapm_dac: case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: @@ -772,6 +773,11 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return widget->inputs; } + /* signal generator */ + if (widget->id == snd_soc_dapm_siggen) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } } list_for_each_entry(path, &widget->sources, list_sink) { @@ -1982,6 +1988,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_out_drv: case snd_soc_dapm_input: case snd_soc_dapm_output: + case snd_soc_dapm_siggen: case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_pre: -- 1.7.7.3

2 6

[alsa-devel] [PATCH] ASoC: Provide debug log of accessory status on WM8958
by Mark Brown 30 Nov '11

30 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5320a9e..65e09e8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2946,6 +2946,8 @@ static void wm8958_default_micdet(u16 status, void *data) struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int report; + dev_dbg(codec->dev, "MICDET %x\n", status); + /* If nothing present then clear our statuses */ if (!(status & WM8958_MICD_STS)) { dev_dbg(codec->dev, "Detected open circuit\n"); -- 1.7.7.3

1 0

Re: [alsa-devel] DM365 McBSP as Master
by Appalayagari Sreedhar 30 Nov '11

30 Nov '11

Hi Vladimir, Thank you very much. i will do the same. on further probing and testing. i found that if i open the device O_WRONLY and O_RDONLY mode it works. if i try to open the device using the O_RDWR mode it fails. is there any configuration settings i need to check. -Sreedhar. On 10/17/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: > Hi Sreedhar, > > You should always CC the alsa-devel channel list for any alsa related > questions instead of writing directly to TI support and me. But you can > CC both me and TI people. > In that case your questions will not go unanswered. > > > Appalayagari Sreedhar wrote: >> Hi Vladimir, >> >> I need your help to solve one issue. >> >> I have developed a audio driver in ALSA model, the audio ic is >> interfaced with the TI DM365 processor MCBSP ports. I am working with >> the 2.6.32 kernel. >> >> I am able to play the audio using the aplay command. but when i try >> to open the device using /dev/dsp i am getting the -ve value. it fails >> to open the device. >> > I believe that you've enabled OSS emulation for alsa drivers: > CONFIG_SND_PCM_OSS=y > and read carefully Documentation/sound/alsa/OSS-Emulation.txt guide. > > Here you didn't mention how you are trying to open /dev/dsp and the > error log sample. > If you'll read the mentioned guide then you'll find some hints about PCM > mode changes via proc interface. Probably that will shine some light on > your issue. > >> when i check the >> >> >>>> ls /dev/dsp >>>> >> >> it is available in the device directory. >> >> I have enabled the Sound OSS related options in the kernel menuconfig. >> with the same options i am able to open the device for the tlv320aic3x >> device. >> >> Could you please suggest where i am doing wrong? and help to solve >> this problem. >> >> One more observation : >> >> after doing aplay i am able to open the device using the /dev/dsp. >> >> if try to open the device using the /dev/dsp before aplay i am getting >> the error. >> >> could you please tell me where i am doing. >> > It seems that aplay configures the codec and mcbsp interface calling > it's TRIGGER_START but your own application does not. Once configured > the pcm streams can pass normally using /dev/dsp. Not usre but seems > that the problem in inability to call TRIGGER. > > Regards, > Vladimir >> I have posted the queries in ti e2e website, please check them below. >> i did not receive any proper response till now so i am contacting you. >> >> http://e2e.ti.com/support/embedded/f/354/p/138037/504805.aspx >> >> Thank you in advance. >> Sreedhar. >> >> On 9/23/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: >> >>> Hi Sreedhar, >>> >>> Appalayagari Sreedhar wrote: >>> >>>> Hi Vladimir Barinov, >>>> >>>> Now I am able to make it work McBSP as master and also as slave. >>>> We decided to configure the McBSP as Slave and Codec as Master. >>>> >>>> I have one question, Since the Clocks are generated by the Codec. >>>> will there be any Channel Swap in the audio? >>>> >>>> If so how to handle this channel swap mechanism. >>>> >>>> >>> Yes, It possible to get channels swap on any external codec. >>> Unfortunately I don't know any strong method for this issue. I just >>> experience similar problem on TI's PMIC audio and get rid from EVERY >>> excessive clock in system to be able to fix it. I mean you have to stop >>> all related clocks in proper sequence during audio pause/stop. When I >>> did it the channel swap was fixed. >>> >>> Regards, >>> Vladimir >>> >>>> Thank you, >>>> Sreedhar. >>>> >>>> >>>> >>>> >>>> On 9/20/11, Appalayagari Sreedhar <sreecha01(a)gmail.com> wrote: >>>> >>>> >>>>> Hi Vladimir Barinov, >>>>> >>>>> Thank you very much. I will check as you suggested. >>>>> Thanks once again you are only person from whom i got the support since >>>>> 5 >>>>> days. >>>>> >>>>> Thanks once again. >>>>> >>>>> Thank you, >>>>> Sreedhar. >>>>> >>>>> On 9/20/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: >>>>> >>>>> >>>>>> Sreedhar, >>>>>> >>>>>> 1) verify that mux registers are configured properly >>>>>> 2) startup with codec as master since the current code tested in this >>>>>> mode and usually msbsp can't provide variety of clocks to match all >>>>>> samplerates. >>>>>> 3) check that you can read/write codec registers with success (i2c >>>>>> works >>>>>> find and the codec is not in reset state - check your h/w) >>>>>> Probe MCLK by scope and find that signal is ON, otherwise you didn't >>>>>> make codec setup (i2c communication or codec doesn't work) >>>>>> >>>>>> After above you can start moving to MCBSP as master if you really >>>>>> need >>>>>> it. >>>>>> >>>>>> Regards, >>>>>> Vladimir >>>>>> >>>>>> Appalayagari Sreedhar wrote: >>>>>> >>>>>> >>>>>>> Hi Vladimir Barinov, >>>>>>> >>>>>>> I am working with DM365 evm board. I have interfaced conexant audio >>>>>>> ic to the McBSP. >>>>>>> >>>>>>> I need to do I2S data transfer. I need to program the McBSP as master >>>>>>> and conexant audio IC as slave. >>>>>>> >>>>>>> i did the macro changes in the davinci-evm.c file in AUDIO_FORMAT >>>>>>> like >>>>>>> this. >>>>>>> >>>>>>> #define AUDIO_FORMAT (SND_SOC_DAIFMT_I2S | \ >>>>>>> SND_SOC_DAIFMT_CBS_CFS) >>>>>>> >>>>>>> I am using the McBSP internal clock for McBSP. I printed the McBSP >>>>>>> registers and found that SCLKME - 0 and CLKSM - 1. by reading the >>>>>>> registers. >>>>>>> >>>>>>> when i run the aplay command i am not viewing any clocks when i probe >>>>>>> the signals using the CRO. >>>>>>> >>>>>>> Could you please tell me where i am doing wrong? >>>>>>> >>>>>>> Please find my queries in the following thread in the ti e2e >>>>>>> comminity. >>>>>>> >>>>>>> http://e2e.ti.com/support/embedded/f/354/p/134788/485628.aspx#485628 >>>>>>> >>>>>>> Thank you, Sreedhar. >>>>>>> >>>>>>> >>>>>>> >>> > >

3 6

[alsa-devel] [PATCH] ASoC: Rename Speyside WM8962 to Tobermory
by Mark Brown 30 Nov '11

30 Nov '11

All the other machine drivers for non-default configurations are named after the relevant audio module so do so for Tobermory also. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- arch/arm/mach-s3c64xx/mach-crag6410.c | 6 +- sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/speyside_wm8962.c | 257 --------------------------------- sound/soc/samsung/tobermory.c | 257 +++++++++++++++++++++++++++++++++ 5 files changed, 264 insertions(+), 264 deletions(-) delete mode 100644 sound/soc/samsung/speyside_wm8962.c create mode 100644 sound/soc/samsung/tobermory.c diff --git a/arch/arm/mach-s3c64xx/mach-crag6410.c b/arch/arm/mach-s3c64xx/mach-crag6410.c index 633f729..4ce4a74 100644 --- a/arch/arm/mach-s3c64xx/mach-crag6410.c +++ b/arch/arm/mach-s3c64xx/mach-crag6410.c @@ -286,8 +286,8 @@ static struct platform_device lowland_device = { .id = -1, }; -static struct platform_device speyside_wm8962_device = { - .name = "speyside-wm8962", +static struct platform_device tobermory_device = { + .name = "tobermory", .id = -1, }; @@ -347,7 +347,7 @@ static struct platform_device *crag6410_devices[] __initdata = { &crag6410_lcd_powerdev, &crag6410_backlight_device, &speyside_device, - &speyside_wm8962_device, + &tobermory_device, &littlemill_device, &lowland_device, &wallvdd_device, diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 9634564..f5a857f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -194,8 +194,8 @@ config SND_SOC_SPEYSIDE select SND_SOC_WM1250_EV1 select SND_SOC_WM0010 -config SND_SOC_SPEYSIDE_WM8962 - tristate "Audio support for Wolfson Speyside with WM8962" +config SND_SOC_TOBERMORY + tristate "Audio support for Wolfson Tobermory" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S select SND_SOC_WM8962 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index c9564e3..9d03beb 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -39,7 +39,7 @@ snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o -snd-soc-speyside-wm8962-objs := speyside_wm8962.o +snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o @@ -62,6 +62,6 @@ obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o -obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o +obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c deleted file mode 100644 index c09648e..0000000 --- a/sound/soc/samsung/speyside_wm8962.c +++ /dev/null @@ -1,257 +0,0 @@ -/* - * Speyside with WM8962 audio support - * - * Copyright 2011 Wolfson Microelectronics - * - * 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; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/jack.h> -#include <linux/gpio.h> -#include <linux/module.h> - -#include "../codecs/wm8962.h" - -static int sample_rate = 44100; - -static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, 32768, - sample_rate * 512); - if (ret < 0) - pr_err("Failed to start FLL: %d\n", ret); - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, - sample_rate * 512, - SND_SOC_CLOCK_IN); - if (ret < 0) { - pr_err("Failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - 32768, SND_SOC_CLOCK_IN); - if (ret < 0) { - pr_err("Failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - dapm->bias_level = level; - - return 0; -} - -static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - sample_rate = params_rate(params); - - return 0; -} - -static struct snd_soc_ops speyside_wm8962_ops = { - .hw_params = speyside_wm8962_hw_params, -}; - -static struct snd_soc_dai_link speyside_wm8962_dai[] = { - { - .name = "CPU", - .stream_name = "CPU", - .cpu_dai_name = "samsung-i2s.0", - .codec_dai_name = "wm8962", - .platform_name = "samsung-audio", - .codec_name = "wm8962.1-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, - .ops = &speyside_wm8962_ops, - }, -}; - -static const struct snd_kcontrol_new controls[] = { - SOC_DAPM_PIN_SWITCH("Main Speaker"), - SOC_DAPM_PIN_SWITCH("DMIC"), -}; - -static struct snd_soc_dapm_widget widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - - SND_SOC_DAPM_MIC("DMIC", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - - SND_SOC_DAPM_SPK("Main Speaker", NULL), -}; - -static struct snd_soc_dapm_route audio_paths[] = { - { "Headphone", NULL, "HPOUTL" }, - { "Headphone", NULL, "HPOUTR" }, - - { "Main Speaker", NULL, "SPKOUTL" }, - { "Main Speaker", NULL, "SPKOUTR" }, - - { "Headset Mic", NULL, "MICBIAS" }, - { "IN4L", NULL, "Headset Mic" }, - { "IN4R", NULL, "Headset Mic" }, - - { "AMIC", NULL, "MICBIAS" }, - { "IN1L", NULL, "AMIC" }, - { "IN1R", NULL, "AMIC" }, - - { "DMIC", NULL, "MICBIAS" }, - { "DMICDAT", NULL, "DMIC" }, -}; - -static struct snd_soc_jack speyside_wm8962_headset; - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin speyside_wm8962_headset_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static int speyside_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - 32768, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &speyside_wm8962_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&speyside_wm8962_headset, - ARRAY_SIZE(speyside_wm8962_headset_pins), - speyside_wm8962_headset_pins); - if (ret) - return ret; - - wm8962_mic_detect(codec, &speyside_wm8962_headset); - - return 0; -} - -static struct snd_soc_card speyside_wm8962 = { - .name = "Speyside WM8962", - .dai_link = speyside_wm8962_dai, - .num_links = ARRAY_SIZE(speyside_wm8962_dai), - - .set_bias_level = speyside_wm8962_set_bias_level, - .set_bias_level_post = speyside_wm8962_set_bias_level_post, - - .controls = controls, - .num_controls = ARRAY_SIZE(controls), - .dapm_widgets = widgets, - .num_dapm_widgets = ARRAY_SIZE(widgets), - .dapm_routes = audio_paths, - .num_dapm_routes = ARRAY_SIZE(audio_paths), - .fully_routed = true, - - .late_probe = speyside_wm8962_late_probe, -}; - -static __devinit int speyside_wm8962_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &speyside_wm8962; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; - } - - return 0; -} - -static int __devexit speyside_wm8962_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - -static struct platform_driver speyside_wm8962_driver = { - .driver = { - .name = "speyside-wm8962", - .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, - }, - .probe = speyside_wm8962_probe, - .remove = __devexit_p(speyside_wm8962_remove), -}; - -module_platform_driver(speyside_wm8962_driver); - -MODULE_DESCRIPTION("Speyside WM8962 audio support"); -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:speyside-wm8962"); diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c new file mode 100644 index 0000000..6f91c65 --- /dev/null +++ b/sound/soc/samsung/tobermory.c @@ -0,0 +1,257 @@ +/* + * Tobermory audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * 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; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/gpio.h> +#include <linux/module.h> + +#include "../codecs/wm8962.h" + +static int sample_rate = 44100; + +static int tobermory_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 32768, + sample_rate * 512); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, + sample_rate * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + return 0; +} + +static int tobermory_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int tobermory_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + + return 0; +} + +static struct snd_soc_ops tobermory_ops = { + .hw_params = tobermory_hw_params, +}; + +static struct snd_soc_dai_link tobermory_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm8962", + .platform_name = "samsung-audio", + .codec_name = "wm8962.1-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &tobermory_ops, + }, +}; + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Main Speaker"), + SOC_DAPM_PIN_SWITCH("DMIC"), +}; + +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_MIC("DMIC", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + + SND_SOC_DAPM_SPK("Main Speaker", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "Headphone", NULL, "HPOUTL" }, + { "Headphone", NULL, "HPOUTR" }, + + { "Main Speaker", NULL, "SPKOUTL" }, + { "Main Speaker", NULL, "SPKOUTR" }, + + { "Headset Mic", NULL, "MICBIAS" }, + { "IN4L", NULL, "Headset Mic" }, + { "IN4R", NULL, "Headset Mic" }, + + { "AMIC", NULL, "MICBIAS" }, + { "IN1L", NULL, "AMIC" }, + { "IN1R", NULL, "AMIC" }, + + { "DMIC", NULL, "MICBIAS" }, + { "DMICDAT", NULL, "DMIC" }, +}; + +static struct snd_soc_jack tobermory_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin tobermory_headset_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int tobermory_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &tobermory_headset); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&tobermory_headset, + ARRAY_SIZE(tobermory_headset_pins), + tobermory_headset_pins); + if (ret) + return ret; + + wm8962_mic_detect(codec, &tobermory_headset); + + return 0; +} + +static struct snd_soc_card tobermory = { + .name = "Tobermory", + .dai_link = tobermory_dai, + .num_links = ARRAY_SIZE(tobermory_dai), + + .set_bias_level = tobermory_set_bias_level, + .set_bias_level_post = tobermory_set_bias_level_post, + + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), + .fully_routed = true, + + .late_probe = tobermory_late_probe, +}; + +static __devinit int tobermory_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &tobermory; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit tobermory_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver tobermory_driver = { + .driver = { + .name = "tobermory", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tobermory_probe, + .remove = __devexit_p(tobermory_remove), +}; + +module_platform_driver(tobermory_driver); + +MODULE_DESCRIPTION("Tobermory audio support"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:tobermory"); -- 1.7.7.3

1 0

Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Takashi Iwai 30 Nov '11

30 Nov '11

At Tue, 29 Nov 2011 15:15:01 -0600, Pierre-Louis Bossart wrote: > > > > 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. > > Power consumption will be more optimized if we push bytes without looking > for block boundaries on the host. But if your hardware requires parsing on > the host, there's no impact on the API. You'd just write block-by-block and > specify the block length instead of fixed-size value. Well, my question is much more simple. In the patch, there are checks of the stream direction in snd_compr_{fragment|frame}_elapsed() functions. +void snd_compr_fragment_elapsed(struct snd_compr_stream *stream) +{ + pr_debug("fragment elapsed called\n"); + if (stream->direction != SNDRV_DEVICE_TYPE_COMPR_PLAYBACK) + return; + wake_up(&stream->runtime->sleep); +} +EXPORT_SYMBOL_GPL(snd_compr_fragment_elapsed); + +void snd_compr_frame_elapsed(struct snd_compr_stream *stream) +{ + if (stream->direction != SNDRV_DEVICE_TYPE_COMPR_CAPTURE) + return; + wake_up(&stream->runtime->sleep); +} +EXPORT_SYMBOL_GPL(snd_compr_frame_elapsed); And I wonder why this restriction *must* be present in the API-level implementation. And, furthermore, I see no reason to have this individual functions. Why simply not an inline function like static inline void snd_compr_stream_elapsed(struct snd_compr_stream *stream) { wake_up(&stream->runtime->sleep); } ?? Then you can reduce two exported symbols and less stack usages. Takashi

2 3

[alsa-devel] ASoC updates for 3.2
by Mark Brown 30 Nov '11

30 Nov '11

The following changes since commit fc8e6e8668e74fbf8e00d6e143d7f43b20f73f32: ASoC: Supply dcs_codes for newer WM1811 revisions (2011-11-28 23:18:38 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 fixing some bugs in the SigmaDSP code. It's in the firmware subsystem but that's orphaned and changes in -next move the code over to the ASoC tree as these DSPs are specialist DSPs for audio use. Lars-Peter Clausen (3): firmware: Sigma: Prevent out of bounds memory access firmware: Sigma: Skip header during CRC generation firmware: Sigma: Fix endianess issues drivers/firmware/sigma.c | 81 +++++++++++++++++++++++++++++++++------------- include/linux/sigma.h | 13 ++----- 2 files changed, 62 insertions(+), 32 deletions(-)

2 1

Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Mark Brown 30 Nov '11

30 Nov '11

On Tue, Nov 29, 2011 at 03:15:01PM -0600, Pierre-Louis Bossart wrote: > > 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. > Power consumption will be more optimized if we push bytes without looking > for block boundaries on the host. But if your hardware requires parsing on > the host, there's no impact on the API. You'd just write block-by-block and > specify the block length instead of fixed-size value. We've got a block based driver in development for this API - it's not had a problem interoperating thus far. So long as we can report how much data we've actually consumed I'd not anticipate any problems.

1 0

[alsa-devel] [PATCH V3] ASoC: Tegra I2S: Add device tree binding
by Stephen Warren 30 Nov '11

30 Nov '11

Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v3: Fix compile warning; s/&of_dma/of_dma/. v2: * Modified binding to include the DMA controller's phandle. This value isn't used by the driver, but is useful to correctly represent the HW. * Renamed dma-channel property to dma-request-selector to better match what it describes. Added missing "nvidia," prefix to it. .../devicetree/bindings/sound/tegra20-i2s.txt | 17 ++++++++++++ sound/soc/tegra/tegra_i2s.c | 27 ++++++++++++++++--- 2 files changed, 39 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra20-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/tegra20-i2s.txt b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt new file mode 100644 index 0000000..0df2b5c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt @@ -0,0 +1,17 @@ +NVIDIA Tegra 20 I2S controller + +Required properties: +- compatible : "nvidia,tegra20-i2s" +- reg : Should contain I2S registers location and length +- interrupts : Should contain I2S interrupt +- nvidia,dma-request-selector : The Tegra DMA controller's phandle and + request selector for this I2S controller + +Example: + +i2s@70002800 { + compatible = "nvidia,tegra20-i2s"; + reg = <0x70002800 0x200>; + interrupts = < 45 >; + nvidia,dma-request-selector = < &apbdma 2 >; +}; diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index ca4d0c0..33509de 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -36,6 +36,7 @@ #include <linux/seq_file.h> #include <linux/slab.h> #include <linux/io.h> +#include <linux/of.h> #include <mach/iomap.h> #include <sound/core.h> #include <sound/pcm.h> @@ -331,6 +332,8 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) { struct tegra_i2s * i2s; struct resource *mem, *memregion, *dmareq; + u32 of_dma[2]; + u32 dma_ch; int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); @@ -360,9 +363,16 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + dma_ch = of_dma[1]; + } else { + dma_ch = dmareq->start; } memregion = devm_request_mem_region(&pdev->dev, mem->start, @@ -383,12 +393,12 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; i2s->capture_dma_data.wrap = 4; i2s->capture_dma_data.width = 32; - i2s->capture_dma_data.req_sel = dmareq->start; + i2s->capture_dma_data.req_sel = dma_ch; i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; i2s->playback_dma_data.wrap = 4; i2s->playback_dma_data.width = 32; - i2s->playback_dma_data.req_sel = dmareq->start; + i2s->playback_dma_data.req_sel = dma_ch; i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; @@ -422,10 +432,16 @@ static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-i2s", }, + {}, +}; + static struct platform_driver tegra_i2s_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = tegra_i2s_of_match, }, .probe = tegra_i2s_platform_probe, .remove = __devexit_p(tegra_i2s_platform_remove), @@ -436,3 +452,4 @@ MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); MODULE_DESCRIPTION("Tegra I2S ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); -- 1.7.0.4

2 1

[alsa-devel] Tests of the hda-jack branch
by David Henningsson 30 Nov '11

30 Nov '11

Hi Takashi, I've done some more tests of your branch now, and the new names seems to work for my machine here. I'm attaching two patches - one adds a "device index" for mapping against pcm devices on HDMI, the other one is a backport for the driver-build tree. What are your thoughts on how to proceed, when do you think this is ready to be merged into the topic/hda branch? // David

5 10