September 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Fix another cs4270 error path
by Jean Delvare 29 Sep '08

29 Sep '08

Conversion to new-style i2c driver missed the error path of the probe function. Fix it. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) --- linux-2.6.27-rc7.orig/sound/soc/codecs/cs4270.c 2008-09-27 17:45:47.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/codecs/cs4270.c 2008-09-27 18:10:52.000000000 +0200 @@ -610,17 +610,12 @@ static int cs4270_i2c_probe(struct i2c_c return 0; error: - if (codec->control_data) { - i2c_detach_client(i2c_client); - codec->control_data = NULL; - } + codec->control_data = NULL; kfree(codec->reg_cache); codec->reg_cache = NULL; codec->reg_cache_size = 0; - kfree(i2c_client); - return ret; } -- Jean Delvare

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[alsa-devel] [PATCH 1/5] [RFC] ALSA ASoC driver for TLVaic23b audio codec
by Arun KS 29 Sep '08

29 Sep '08

ASOC Audio driver for TLVaic23b audio codec Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 634 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 143 +++++++++ 4 files changed, 783 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b1a5eed..7be0b37 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,3 +55,7 @@ config SND_SOC_TWL4030 tristate depends on SND_SOC && TWL4030_CORE +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a519ced..adb5aa6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-tlv320aic23-objs := tlv320aic23.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -25,3 +26,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 0000000..eefeb2d --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,634 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/tlv320aic23.c by Vladimir Barinov + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#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/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AUDIO_NAME "aic23" +#define AIC23_VERSION "0.1" + +/* codec private data */ +struct aic23_priv { + unsigned int sysclk; + int master; +}; + +/* + * AIC23 register cache + */ +static const u16 aic23_reg[AIC23_CACHEREGNUM] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read aic23 register cache + */ +static inline unsigned int aic23_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AIC23_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write aic23 register cache + */ +static inline void aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AIC23_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the aic23 register space + */ +static int aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "Invalid register R%d\n", reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, (u8) data, + (u8) (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "I2C: cannot write %03x to register R%d\n", value, reg); + + return -EIO; +} + +static const struct snd_kcontrol_new aic23_snd_controls[] = { + SOC_DOUBLE_R("PCM Playback Volume", LEFT_CHANNEL_VOLUME_ADDR, + RIGHT_CHANNEL_VOLUME_ADDR, 0, 0x7f, 0), + SOC_SINGLE("PCM Playback Switch", DIGITAL_AUDIO_CONTROL_ADDR, 3, 0x01, + 1), + SOC_DOUBLE_R("Line Input Mute", LEFT_LINE_VOLUME_ADDR, + RIGHT_LINE_VOLUME_ADDR, 7, 0x01, 0), + SOC_DOUBLE_R("Line Input Volume", LEFT_LINE_VOLUME_ADDR, + RIGHT_LINE_VOLUME_ADDR, 0, 0x1f, 0), + SOC_SINGLE("Mic Mute", ANALOG_AUDIO_CONTROL_ADDR, 1, 0x01, 0), + SOC_SINGLE("Mic Booster Switch", ANALOG_AUDIO_CONTROL_ADDR, 0, 0x01, 0), + +}; + +/* add non dapm controls */ +static int aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} +static const char *aic23_rec_src[] = { "Line", "Mic" }; + +static const struct soc_enum aic23_enum[] = { + SOC_ENUM_SINGLE(ANALOG_AUDIO_CONTROL_ADDR, 2, 2, aic23_rec_src), +}; + +static const struct snd_kcontrol_new aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Route", aic23_enum[0]); + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new aic23_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", POWER_DOWN_CONTROL_ADDR, 0, 1, 1), + SOC_DAPM_SINGLE("Mic Switch", POWER_DOWN_CONTROL_ADDR, 1, 1, 1), +}; + +static const struct snd_soc_dapm_widget aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", POWER_DOWN_CONTROL_ADDR, 3, 1), + SND_SOC_DAPM_PGA("OUT", POWER_DOWN_CONTROL_ADDR, 4, 1, NULL, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", POWER_DOWN_CONTROL_ADDR, 2, 1), + SND_SOC_DAPM_MIXER("PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic23_pga_mixer_controls[0], + ARRAY_SIZE(aic23_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &aic23_rec_src_mux_controls), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"OUT", NULL, "DAC"}, + + {"LHPOUT", NULL, "OUT"}, + {"RHPOUT", NULL, "OUT"}, + + {"LOUT", NULL, "OUT"}, + {"ROUT", NULL, "OUT"}, + + {"Capture Source", "Line", "LLINEIN"}, + {"Capture Source", "Line", "RLINEIN"}, + + {"Capture Source", "Mic", "MICIN"}, + + {"PGA Mixer", "Line Switch", "Capture Source"}, + {"PGA Mixer", "Mic Switch", "Capture Source"}, + + {"ADC", NULL, "PGA Mixer"}, + +}; + +/* aic23 related */ +static const struct aic23_samplerate_reg_info + rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic23_dapm_widgets, + ARRAY_SIZE(aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + aic23_read_reg_cache(codec, + DIGITAL_AUDIO_FORMAT_ADDR) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((rate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(rate_reg_info))) { + count++; + } + + data = (rate_reg_info[count].divider << CLKIN_SHIFT) | + (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON; + + aic23_write(codec, SAMPLE_RATE_CONTROL_ADDR, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg); + + return 0; +} + +static int aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = + aic23_read_reg_cache(codec, + DIGITAL_AUDIO_CONTROL_ADDR) & ~DACM_MUTE; + + if (mute) + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg | DACM_MUTE); + + else + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg); + + return 0; +} + +static int aic23_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 aic23_priv *aic23 = codec->private_data; + + aic23->sysclk = freq; + return 0; +} + +static int aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic23_priv *aic23 = codec->private_data; + u16 iface_reg; + + iface_reg = + aic23_read_reg_cache(codec, DIGITAL_AUDIO_FORMAT_ADDR) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic23->master = 1; + iface_reg |= MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic23->master = 0; + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= FOR_LJUST; + break; + default: + return -EINVAL; + + } + + aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg); + + return 0; +} + +static int aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, + reg & (~DEVICE_POWER_OFF)); + /* Activate interface */ + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg | ACT_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, + reg | DEVICE_POWER_OFF); + /* Deactivate interface */ + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg & (~ACT_ON)); + + break; + case SND_SOC_BIAS_OFF: + break; + } + + codec->bias_level = level; + + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai aic23_dai = { + .name = "aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .hw_params = aic23_hw_params, + }, + .dai_ops = { + .digital_mute = aic23_mute, + .set_sysclk = aic23_set_dai_sysclk, + .set_fmt = aic23_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(aic23_dai); + +static int aic23_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(aic23_reg); i++) { + data[0] = i; + data[1] = cache[i]; + codec->hw_write(codec->control_data, data, 2); + } + + aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "aic23"; + codec->owner = THIS_MODULE; + codec->read = aic23_read_reg_cache; + codec->write = aic23_write; + codec->set_bias_level = aic23_set_bias_level; + codec->dai = &aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(aic23_reg); + codec->reg_cache = kmemdup(aic23_reg, sizeof(aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + aic23_write(codec, RESET_CONTROL_ADDR, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to create pcms\n"); + goto pcm_err; + } + + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K); + + /* Unmute input */ + reg = aic23_read_reg_cache(codec, LEFT_LINE_VOLUME_ADDR); + aic23_write(codec, LEFT_LINE_VOLUME_ADDR, + (reg & (~LIM_MUTED)) | (LRS_ENABLED)); + reg = aic23_read_reg_cache(codec, RIGHT_LINE_VOLUME_ADDR); + aic23_write(codec, RIGHT_LINE_VOLUME_ADDR, + (reg & (~LIM_MUTED)) | LRS_ENABLED); + reg = aic23_read_reg_cache(codec, ANALOG_AUDIO_CONTROL_ADDR); + aic23_write(codec, ANALOG_AUDIO_CONTROL_ADDR, + (reg | DAC_SELECTED) & (~MICM_MUTED) & (~BYPASS_ON)); + + /* Default output volume */ + aic23_write(codec, LEFT_CHANNEL_VOLUME_ADDR, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + aic23_write(codec, RIGHT_CHANNEL_VOLUME_ADDR, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + + /* off, with power on */ + aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + aic23_add_controls(codec); + aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit aic23_i2c_remove(struct i2c_client *i2c) +{ + + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlvaic23b_id[] = { + {"tlvaic23b", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlvaic23b_id); + +static struct i2c_driver aic23b_i2c_driver = { + .driver = { + .name = "tlvaic23b", + }, + .probe = aic23_codec_probe, + .remove = __exit_p(aic23_i2c_remove), + .id_table = tlvaic23b_id, +}; + +#endif + +static int aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct aic23_priv *aic23; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + aic23 = kzalloc(sizeof(struct aic23_priv), GFP_KERNEL); + if (aic23 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = aic23; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&aic23b_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#else + /* Add other interfaces here */ +#endif + return ret; +} + +static int aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + /* power down chip */ + if (codec->control_data) + aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&aic23b_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_aic23 = { + .probe = aic23_probe, + .remove = aic23_remove, + .suspend = aic23_suspend, + .resume = aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 0000000..9c3808c --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,143 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks(a)mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AIC23_H +#define _AIC23_H + +#define AIC23_CACHEREGNUM 16 +/* Codec TLV320AIC23 */ +#define LEFT_LINE_VOLUME_ADDR 0x00 +#define RIGHT_LINE_VOLUME_ADDR 0x01 +#define LEFT_CHANNEL_VOLUME_ADDR 0x02 +#define RIGHT_CHANNEL_VOLUME_ADDR 0x03 +#define ANALOG_AUDIO_CONTROL_ADDR 0x04 +#define DIGITAL_AUDIO_CONTROL_ADDR 0x05 +#define POWER_DOWN_CONTROL_ADDR 0x06 +#define DIGITAL_AUDIO_FORMAT_ADDR 0x07 +#define SAMPLE_RATE_CONTROL_ADDR 0x08 +#define DIGITAL_INTERFACE_ACT_ADDR 0x09 +#define RESET_CONTROL_ADDR 0x0F + +/* Left (right) line input volume control register */ +#define LRS_ENABLED 0x0100 +#define LIM_MUTED 0x0080 +#define LIV_DEFAULT 0x0017 +#define LIV_MAX 0x001f +#define LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define LZC_ON 0x0080 +#define LHV_DEFAULT 0x0079 +#define LHV_MAX 0x007f +#define LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define STA_REG(x) ((x)<<6) +#define STE_ENABLED 0x0020 +#define DAC_SELECTED 0x0010 +#define BYPASS_ON 0x0008 +#define INSEL_MIC 0x0004 +#define MICM_MUTED 0x0002 +#define MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define DACM_MUTE 0x0008 +#define DEEMP_32K 0x0002 +#define DEEMP_44K 0x0004 +#define DEEMP_48K 0x0006 +#define ADCHP_ON 0x0001 + +/* Power control down register */ +#define DEVICE_POWER_OFF 0x0080 +#define CLK_OFF 0x0040 +#define OSC_OFF 0x0020 +#define OUT_OFF 0x0010 +#define DAC_OFF 0x0008 +#define ADC_OFF 0x0004 +#define MIC_OFF 0x0002 +#define LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define MS_MASTER 0x0040 +#define LRSWAP_ON 0x0020 +#define LRP_ON 0x0010 +#define IWL_16 0x0000 +#define IWL_20 0x0004 +#define IWL_24 0x0008 +#define IWL_32 0x000C +#define FOR_I2S 0x0002 +#define FOR_DSP 0x0003 +#define FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define CLKOUT_HALF 0x0080 +#define CLKIN_HALF 0x0040 +#define BOSR_384fs 0x0002 /* BOSR_272fs when in USB mode */ +#define USB_CLK_ON 0x0001 +#define SR_MASK 0xf +#define CLKOUT_SHIFT 7 +#define CLKIN_SHIFT 6 +#define SR_SHIFT 2 +#define BOSR_SHIFT 1 + +/* Digital interface register */ +#define ACT_ON 0x0001 + +/* Define to set the AIC23 as the master w.r.t McBSP */ +#define AIC23_MASTER + +#define NUMBER_SAMPLE_RATES_SUPPORTED 10 + +/* + * AUDIO related MACROS + */ +#ifndef DEFAULT_BITPERSAMPLE +#define DEFAULT_BITPERSAMPLE 16 +#endif + +#define DEFAULT_SAMPLE_RATE 44100 +#define CODEC_CLOCK 12000000 +#define AUDIO_MCBSP OMAP_MCBSP1 + +#define DEFAULT_OUTPUT_VOLUME 0x70 +#define DEFAULT_INPUT_VOLUME 0x10 /* 0 ==> mute line in */ + +#define OUTPUT_VOLUME_MIN LHV_MIN +#define OUTPUT_VOLUME_MAX LHV_MAX +#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN) +#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX + +#define INPUT_VOLUME_MIN LIV_MIN +#define INPUT_VOLUME_MAX LIV_MAX +#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN) +#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX + +#define SIDETONE_MASK 0x1c0 +#define SIDETONE_0 0x100 +#define SIDETONE_6 0x000 +#define SIDETONE_9 0x040 +#define SIDETONE_12 0x080 +#define SIDETONE_18 0x0c0 + +#define DEFAULT_ANALOG_AUDIO_CONTROL (DAC_SELECTED | STE_ENABLED | \ + INSEL_MIC | MICB_20DB) + +struct aic23_samplerate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +extern struct snd_soc_dai aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_aic23; + +#endif /* _AIC23_H */ + -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH] alsa: make the CS4270 driver a new-style I2C driver
by Timur Tabi 29 Sep '08

29 Sep '08

Update the CS4270 ALSA device driver to use the new-style I2C interface. Starting with the 2.6.27 PowerPC kernel, I2C devices that have entries in the device trees can no longer be probed by old-style I2C drivers. The device tree for Freescale MPC8610 HPCD has included an entry for the CS4270 since 2.6.25, but that entry was previously ignored by the PowerPC I2C subsystem. Since that's no longer the case, the best solution is to update the CS4270 driver to a new-style interface, rather than try to revert the behavior of new PowerPC I2C subsystem. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for 2.6.27 (ASoC V1). It's not needed for 2.6.26, because that kernel uses the older PowerPC I2C code. It's not needed for ASoC V2, because the ASoC V2 CS4270 driver has always been a new-style driver. sound/soc/codecs/cs4270.c | 71 +++++++++----------------------------------- 1 files changed, 15 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 9deb8c7..82d94f0 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -490,34 +490,7 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) #endif -static int cs4270_i2c_probe(struct i2c_adapter *adap, int addr, int kind); - -/* - * Notify the driver that a new I2C bus has been found. - * - * This function is called for each I2C bus in the system. The function - * then asks the I2C subsystem to probe that bus at the addresses on which - * our device (the CS4270) could exist. If a device is found at one of - * those addresses, then our probe function (cs4270_i2c_probe) is called. - */ -static int cs4270_i2c_attach(struct i2c_adapter *adapter) -{ - return i2c_probe(adapter, &addr_data, cs4270_i2c_probe); -} - -static int cs4270_i2c_detach(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - - i2c_detach_client(client); - codec->control_data = NULL; - - kfree(codec->reg_cache); - codec->reg_cache = NULL; - - kfree(client); - return 0; -} +static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *); /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { @@ -525,14 +498,19 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) }; +static const struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "CS4270 I2C", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_CS4270, - .attach_adapter = cs4270_i2c_attach, - .detach_client = cs4270_i2c_detach, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, }; /* @@ -561,11 +539,11 @@ static struct snd_soc_device *cs4270_socdev; * Note: snd_soc_new_pcms() must be called before this function can be called, * because of snd_ctl_add(). */ -static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) +static int cs4270_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = cs4270_socdev; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c_client = NULL; int i; int ret = 0; @@ -578,12 +556,6 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) /* Note: codec_dai->codec is NULL here */ - i2c_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); - if (!i2c_client) { - printk(KERN_ERR "cs4270: could not allocate I2C client\n"); - return -ENOMEM; - } - codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); if (!codec->reg_cache) { printk(KERN_ERR "cs4270: could not allocate register cache\n"); @@ -591,13 +563,6 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) goto error; } - i2c_set_clientdata(i2c_client, codec); - strcpy(i2c_client->name, "CS4270"); - - i2c_client->driver = &cs4270_i2c_driver; - i2c_client->adapter = adapter; - i2c_client->addr = addr; - /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); @@ -612,18 +577,10 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) goto error; } - printk(KERN_INFO "cs4270: found device at I2C address %X\n", addr); + printk(KERN_INFO "cs4270: found device at I2C address %X\n", + i2c_client->addr); printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); - /* Tell the I2C layer a new client has arrived */ - - ret = i2c_attach_client(i2c_client); - if (ret) { - printk(KERN_ERR "cs4270: could not attach codec, " - "I2C address %x, error code %i\n", addr, ret); - goto error; - } - codec->control_data = i2c_client; codec->read = cs4270_read_reg_cache; codec->write = cs4270_i2c_write; @@ -648,6 +605,8 @@ static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) goto error; } + i2c_set_clientdata(i2c_client, codec); + return 0; error: -- 1.5.5

2 3

[alsa-devel] [PATCH] Drop Vladimir Barinov's e-mail address
by Jean Delvare 29 Sep '08

29 Sep '08

Every e-mail I've sent to Vladimir Barinov bounced back to me, so I guess he no longer works at MontaVista. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> --- arch/arm/plat-omap/include/mach/mtd-xip.h | 2 +- drivers/usb/host/ehci-ixp4xx.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 2 +- sound/soc/davinci/davinci-evm.c | 2 +- sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/davinci/davinci-pcm.h | 2 +- 9 files changed, 9 insertions(+), 9 deletions(-) --- linux-2.6.27-rc7.orig/arch/arm/plat-omap/include/mach/mtd-xip.h 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/arch/arm/plat-omap/include/mach/mtd-xip.h 2008-09-27 12:56:40.000000000 +0200 @@ -3,7 +3,7 @@ * * Do not include this file directly. It's included from linux/mtd/xip.h * - * Author: Vladimir Barinov <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * * (c) 2005 MontaVista Software, Inc. This file is licensed under the * terms of the GNU General Public License version 2. This program is --- linux-2.6.27-rc7.orig/drivers/usb/host/ehci-ixp4xx.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/drivers/usb/host/ehci-ixp4xx.c 2008-09-27 12:56:30.000000000 +0200 @@ -1,7 +1,7 @@ /* * IXP4XX EHCI Host Controller Driver * - * Author: Vladimir Barinov <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * * Based on "ehci-fsl.c" by Randy Vinson <rvinson(a)mvista.com> * --- linux-2.6.27-rc7.orig/sound/soc/codecs/tlv320aic3x.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/codecs/tlv320aic3x.c 2008-09-27 12:56:23.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA SoC TLV320AIC3X codec driver * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * Based on sound/soc/codecs/wm8753.c by Liam Girdwood --- linux-2.6.27-rc7.orig/sound/soc/codecs/tlv320aic3x.h 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/codecs/tlv320aic3x.h 2008-09-27 12:56:19.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA SoC TLV320AIC3X codec driver * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-evm.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-evm.c 2008-09-27 12:56:14.000000000 +0200 @@ -1,7 +1,7 @@ /* * ASoC driver for TI DAVINCI EVM platform * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-i2s.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-i2s.c 2008-09-27 12:56:11.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-i2s.h 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-i2s.h 2008-09-27 12:56:07.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-pcm.c 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-pcm.c 2008-09-27 12:56:03.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA PCM interface for the TI DAVINCI processor * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify --- linux-2.6.27-rc7.orig/sound/soc/davinci/davinci-pcm.h 2008-09-27 12:53:06.000000000 +0200 +++ linux-2.6.27-rc7/sound/soc/davinci/davinci-pcm.h 2008-09-27 12:55:59.000000000 +0200 @@ -1,7 +1,7 @@ /* * ALSA PCM interface for the TI DAVINCI processor * - * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Author: Vladimir Barinov * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify -- Jean Delvare

3 2

[alsa-devel] [PATCH] hda: appletv support
by Peter Korsgaard 29 Sep '08

29 Sep '08

The AppleTV needs the same handling as the 24" iMac. Signed-off-by: Peter Korsgaard <jacmet(a)sunsite.dk> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c2ff85d..b0b665e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6656,6 +6656,7 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC885_MACPRO; break; case 0x106b1000: /* iMac 24 */ + case 0x106b2800: /* AppleTV */ board_config = ALC885_IMAC24; break; case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ -- 1.5.6.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Map ALC662 model for Gigabyte 945GCM-S2L
by Herton Ronaldo Krzesinski 29 Sep '08

29 Sep '08

Use 3stack-6ch-dig ALC662 model for Gigabyte 945GCM-S2L motherboard. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff -p -up linux-2.6.26/sound/pci/hda/patch_realtek.c.orig linux-2.6.26/sound/pci/hda/patch_realtek.c --- linux-2.6.26/sound/pci/hda/patch_realtek.c.orig 2008-09-26 23:11:04.000000000 -0300 +++ linux-2.6.26/sound/pci/hda/patch_realtek.c 2008-09-26 23:22:29.000000000 -0300 @@ -15705,6 +15705,8 @@ static struct snd_pci_quirk alc662_cfg_t SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", + ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13),

2 1

[alsa-devel] [PATCH] hda: SPDIF stream muting support
by Matthew Ranostay 29 Sep '08

29 Sep '08

Added support for muting S/DPIF outs using the 'Digital Playback Source' muxs on supporting codecs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..cb419a0 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -138,6 +138,7 @@ struct sigmatel_spec { unsigned int mic_switch: 1; unsigned int alt_switch: 1; unsigned int hp_detect: 1; + unsigned int spdif_mute: 1; /* gpio lines */ unsigned int eapd_mask; @@ -547,10 +548,32 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *smux = &spec->private_smux; unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int err, val; + hda_nid_t nid; - return snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol, + err = snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol, spec->smux_nids[smux_idx], &spec->cur_smux[smux_idx]); + if (err < 0) + return err; + + if (spec->spdif_mute) { + if (smux_idx == 0) + nid = spec->multiout.dig_out_nid; + else + nid = codec->slave_dig_outs[smux_idx - 1]; + if (spec->cur_smux[smux_idx] == smux->num_items - 1) + val = AMP_OUT_MUTE; + if (smux_idx == 0) + nid = spec->multiout.dig_out_nid; + else + nid = codec->slave_dig_outs[smux_idx - 1]; + /* un/mute SPDIF out */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, val); + } + return 0; } static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1228,6 +1251,15 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } if (spec->num_smuxes > 0) { + int wcaps = get_wcaps(codec, spec->multiout.dig_out_nid); + struct hda_input_mux *smux = &spec->private_smux; + /* check for mute support on SPDIF out */ + if (wcaps & AC_WCAP_OUT_AMP) { + smux->items[smux->num_items].label = "Off"; + smux->items[smux->num_items].index = 0; + smux->num_items++; + spec->spdif_mute = 1; + } stac_smux_mixer.count = spec->num_smuxes; err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&stac_smux_mixer, codec)); @@ -4370,7 +4402,6 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); switch (spec->board_config) { case STAC_HP_M4:

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[alsa-devel] [PATCH 5/5] [RFC] Adds I2C board info for tlvaic23b in osk board file
by Arun KS 29 Sep '08

29 Sep '08

Adding I2C board info required for tlvaic23b i2c chip driver Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- arch/arm/mach-omap1/board-osk.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 3e766e4..8ddae82 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -258,8 +258,10 @@ static struct i2c_board_info __initdata osk_i2c_board_info[] = { .platform_data = &tps_board, }, + { + I2C_BOARD_INFO("tlvaic23b", 0x1B), + }, /* TODO when driver support is ready: - * - aic23 audio chip at 0x1a * - on Mistral, 24c04 eeprom at 0x50 * - optionally on Mistral, ov9640 camera sensor at 0x30 */ -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] Compiling an Alsa plugin with Intel compiler.
by Motti Daniel 28 Sep '08

28 Sep '08

Hello people! I'm working on an Alsa plugin for the Xandros Linux distribution my plugin runs very well except it consume about 20% of resources from the target machine (eee pc at 1.6Ghz) I'm trying to compile the same code with Intel Compiler (version 10.1.017) (before that it was compiled with gcc 4.1.2) the plugin compiles well but when I try to play music it tells me that the plugin library (the .so file in /usr/lib/alsa-lib/ is not there. I tried to analyze the symbols in the executable and found that the difference between gcc and Intel compiler is a list of symbols with a suffix __FUNCTION__ or __PRETTY_FUNCTION__ is this the problem? any suggestions? can you tell me what are the minimum alsa requirements from a dynamic library so it can be loaded and run as an Alsa plugin? thanks! Motti Daniel.

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[alsa-devel] GeForce 8200 HD audio seems to work!
by Vedran Miletić 28 Sep '08

28 Sep '08

I just installed Linux on ASROCK GeForce 8200 board. Being a relatively new chipset, I had a couple of issues booting older kernels, finally settling for Debian lenny since it had a CD that booted. I was worried about onboard audio, since it's one of thousands of HD audio solutions, and I expected to have problems with it. To make the long story short: 1.0.16 -> only noise, doesn't sound close to track played 1.0.17 -> sound + lots of noise, but you can recongnize the song that is playing takashi's sound-2.6 -> everything works (including front panel) + HDMI audio seeems to be supported Massive props to whoever made it work. I just have two questions: 1) On RS690 I used to have separate device for RS690/RS780 HDMI (that is HDA ATI and HDA ATI HDMI), on NVIDIA I only have HDA NVIDIA, which has subdevice for HDMI, why? 2) alsamixer says: Card: HDA NVidia | Chip: Generic 10de NVIDIA MCP78 HDMI. Shouldn't chip be "Realtek ALC662 rev1" or something like that? Also, why does it say Generic 10de? Where is that read from? Regards, -- Vedran Miletić

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