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

[alsa-devel] [PATCH] ASoC: twl6040: Return -ENOMEM if create_singlethread_workqueue fails
by Axel Lin 26 Mar '11

26 Mar '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/twl6040.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 482fcdb..255901c 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1629,8 +1629,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->naudint = naudint; priv->workqueue = create_singlethread_workqueue("twl6040-codec"); - if (!priv->workqueue) + if (!priv->workqueue) { + ret = -ENOMEM; goto work_err; + } INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work); -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: tlv320dac33: Restore L/R DAC power control register
by Peter Ujfalusi 26 Mar '11

26 Mar '11

Register 0x40, 0x41 need to be restored after power up, since it contains gain related fields, which affects playback volume. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 93f38b1..6c3735d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -324,6 +324,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_write(codec, DAC33_OUT_AMP_CTRL, dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL)); + dac33_write(codec, DAC33_LDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_LDAC_PWR_CTRL)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_RDAC_PWR_CTRL)); } static inline int dac33_read_id(struct snd_soc_codec *codec) -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: tlv320dac33: Move codec power up to DAPM
by Peter Ujfalusi 26 Mar '11

26 Mar '11

Move the codec power on (in reg 0x01, bit 4) from set_bias_level:SND_SOC_BIAS_ON to a DAPM supply. In this way we can be sure, that all the things within the codec is powered before the external amp is going to be enabled. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 9 ++++++--- 1 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 6c3735d..c109f6d 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -587,6 +587,9 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Right DAC Power", DAC33_RDAC_PWR_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Codec Power", + DAC33_PWR_CTRL, 4, 0, NULL, 0), + SND_SOC_DAPM_PRE("Pre Playback", dac33_playback_event), SND_SOC_DAPM_POST("Post Playback", dac33_playback_event), }; @@ -619,6 +622,9 @@ static const struct snd_soc_dapm_route audio_map[] = { /* output */ {"LEFT_LO", NULL, "Output Left Amplifier"}, {"RIGHT_LO", NULL, "Output Right Amplifier"}, + + {"LEFT_LO", NULL, "Codec Power"}, + {"RIGHT_LO", NULL, "Codec Power"}, }; static int dac33_add_widgets(struct snd_soc_codec *codec) @@ -636,13 +642,10 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { case SND_SOC_BIAS_ON: - if (!dac33->substream) - dac33_soft_power(codec, 1); break; case SND_SOC_BIAS_PREPARE: break; -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: ad183x: add support for more multiple parts
by Mike Frysinger 26 Mar '11

26 Mar '11

From: Barry Song <barry.song(a)analog.com> This extends the AD1836 codec driver to also support the parts: - AD1835A - AD1837A - AD1838A - AD1839A Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/ad1836.c | 338 -------------------------------- sound/soc/codecs/ad1836.h | 63 ------ sound/soc/codecs/ad183x.c | 474 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad183x.h | 57 ++++++ 6 files changed, 535 insertions(+), 405 deletions(-) delete mode 100644 sound/soc/codecs/ad1836.c delete mode 100644 sound/soc/codecs/ad1836.h create mode 100644 sound/soc/codecs/ad183x.c create mode 100644 sound/soc/codecs/ad183x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6943e24..dc6d253 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -13,7 +13,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS - select SND_SOC_AD1836 if SPI_MASTER + select SND_SOC_AD183X if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_ADS117X @@ -115,7 +115,7 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC -config SND_SOC_AD1836 +config SND_SOC_AD183X tristate config SND_SOC_AD193X diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 379bc55..44d3278 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,6 +1,6 @@ snd-soc-88pm860x-objs := 88pm860x-codec.o snd-soc-ac97-objs := ac97.o -snd-soc-ad1836-objs := ad1836.o +snd-soc-ad183x-objs := ad183x.o snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o @@ -87,7 +87,7 @@ snd-soc-wm9090-objs := wm9090.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o -obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o +obj-$(CONFIG_SND_SOC_AD183X) += snd-soc-ad183x.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c deleted file mode 100644 index ab63d52..0000000 --- a/sound/soc/codecs/ad1836.c +++ /dev/null @@ -1,338 +0,0 @@ -/* - * File: sound/soc/codecs/ad1836.c - * Author: Barry Song <Barry.Song(a)analog.com> - * - * Created: Aug 04 2009 - * Description: Driver for AD1836 sound chip - * - * Modified: - * Copyright 2009 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * 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 <linux/init.h> -#include <linux/slab.h> -#include <linux/module.h> -#include <linux/kernel.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> -#include <sound/tlv.h> -#include <linux/spi/spi.h> -#include "ad1836.h" - -/* codec private data */ -struct ad1836_priv { - enum snd_soc_control_type control_type; - void *control_data; -}; - -/* - * AD1836 volume/mute/de-emphasis etc. controls - */ -static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; - -static const struct soc_enum ad1836_deemp_enum = - SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp); - -static const struct snd_kcontrol_new ad1836_snd_controls[] = { - /* DAC volume control */ - SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL, - AD1836_DAC_R1_VOL, 0, 0x3FF, 0), - SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL, - AD1836_DAC_R2_VOL, 0, 0x3FF, 0), - SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL, - AD1836_DAC_R3_VOL, 0, 0x3FF, 0), - - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE, - AD1836_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE, - AD1836_ADCR2_MUTE, 1, 1), - - /* DAC switch control */ - SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE, - AD1836_DACR1_MUTE, 1, 1), - SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE, - AD1836_DACR2_MUTE, 1, 1), - SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE, - AD1836_DACR3_MUTE, 1, 1), - - /* ADC high-pass filter */ - SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1, - AD1836_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum), -}; - -static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1, - AD1836_DAC_POWERDOWN, 1), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1, - AD1836_ADC_POWERDOWN, 1, NULL, 0), - SND_SOC_DAPM_OUTPUT("DAC1OUT"), - SND_SOC_DAPM_OUTPUT("DAC2OUT"), - SND_SOC_DAPM_OUTPUT("DAC3OUT"), - SND_SOC_DAPM_INPUT("ADC1IN"), - SND_SOC_DAPM_INPUT("ADC2IN"), -}; - -static const struct snd_soc_dapm_route audio_paths[] = { - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", "DAC1 Switch", "DAC" }, - { "DAC2OUT", "DAC2 Switch", "DAC" }, - { "DAC3OUT", "DAC3 Switch", "DAC" }, - { "ADC", "ADC1 Switch", "ADC1IN" }, - { "ADC", "ADC2 Switch", "ADC2IN" }, -}; - -/* - * DAI ops entries - */ - -static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) -{ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - /* at present, we support adc aux mode to interface with - * blackfin sport tdm mode - */ - case SND_SOC_DAIFMT_DSP_A: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_IF: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - /* ALCLK,ABCLK are both output, AD1836 can only be master */ - case SND_SOC_DAIFMT_CBM_CFM: - break; - default: - return -EINVAL; - } - - return 0; -} - -static int ad1836_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - int word_len = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - - /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - word_len = 3; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - word_len = 1; - break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: - word_len = 0; - break; - } - - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, - AD1836_DAC_WORD_LEN_MASK, word_len); - - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, - AD1836_ADC_WORD_LEN_MASK, word_len); - - return 0; -} - -#ifdef CONFIG_PM -static int ad1836_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) -{ - /* reset clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} - -static int ad1836_soc_resume(struct snd_soc_codec *codec) -{ - /* restore clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 |= AD1836_ADC_AUX; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} -#else -#define ad1836_soc_suspend NULL -#define ad1836_soc_resume NULL -#endif - -static struct snd_soc_dai_ops ad1836_dai_ops = { - .hw_params = ad1836_hw_params, - .set_fmt = ad1836_set_dai_fmt, -}; - -/* codec DAI instance */ -static struct snd_soc_dai_driver ad1836_dai = { - .name = "ad1836-hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 2, - .channels_max = 6, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .ops = &ad1836_dai_ops, -}; - -static int ad1836_probe(struct snd_soc_codec *codec) -{ - struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret = 0; - - codec->control_data = ad1836->control_data; - ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", - ret); - return ret; - } - - /* default setting for ad1836 */ - /* de-emphasis: 48kHz, power-on dac */ - snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300); - /* unmute dac channels */ - snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0); - /* high-pass filter enable, power-on adc */ - snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); - /* unmute adc channles, adc aux mode */ - snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); - /* left/right diff:PGA/MUX */ - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); - /* volume */ - snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF); - - snd_soc_add_controls(codec, ad1836_snd_controls, - ARRAY_SIZE(ad1836_snd_controls)); - snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets, - ARRAY_SIZE(ad1836_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); - - return ret; -} - -/* power down chip */ -static int ad1836_remove(struct snd_soc_codec *codec) -{ - /* reset clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); -} - -static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { - .probe = ad1836_probe, - .remove = ad1836_remove, - .suspend = ad1836_soc_suspend, - .resume = ad1836_soc_resume, - .reg_cache_size = AD1836_NUM_REGS, - .reg_word_size = sizeof(u16), -}; - -static int __devinit ad1836_spi_probe(struct spi_device *spi) -{ - struct ad1836_priv *ad1836; - int ret; - - ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL); - if (ad1836 == NULL) - return -ENOMEM; - - spi_set_drvdata(spi, ad1836); - ad1836->control_data = spi; - ad1836->control_type = SND_SOC_SPI; - - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ad1836, &ad1836_dai, 1); - if (ret < 0) - kfree(ad1836); - return ret; -} - -static int __devexit ad1836_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); - return 0; -} - -static struct spi_driver ad1836_spi_driver = { - .driver = { - .name = "ad1836-codec", - .owner = THIS_MODULE, - }, - .probe = ad1836_spi_probe, - .remove = __devexit_p(ad1836_spi_remove), -}; - -static int __init ad1836_init(void) -{ - int ret; - - ret = spi_register_driver(&ad1836_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n", - ret); - } - - return ret; -} -module_init(ad1836_init); - -static void __exit ad1836_exit(void) -{ - spi_unregister_driver(&ad1836_spi_driver); -} -module_exit(ad1836_exit); - -MODULE_DESCRIPTION("ASoC ad1836 driver"); -MODULE_AUTHOR("Barry Song <21cnbao(a)gmail.com>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h deleted file mode 100644 index 8455967..0000000 --- a/sound/soc/codecs/ad1836.h +++ /dev/null @@ -1,63 +0,0 @@ -/* - * File: sound/soc/codecs/ad1836.h - * Based on: - * Author: Barry Song <Barry.Song(a)analog.com> - * - * Created: Aug 04, 2009 - * Description: definitions for AD1836 registers - * - * Modified: - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * 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. - */ - -#ifndef __AD1836_H__ -#define __AD1836_H__ - -#define AD1836_DAC_CTRL1 0 -#define AD1836_DAC_POWERDOWN 2 -#define AD1836_DAC_SERFMT_MASK 0xE0 -#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) -#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) -#define AD1836_DAC_WORD_LEN_MASK 0x18 - -#define AD1836_DAC_CTRL2 1 -#define AD1836_DACL1_MUTE 0 -#define AD1836_DACR1_MUTE 1 -#define AD1836_DACL2_MUTE 2 -#define AD1836_DACR2_MUTE 3 -#define AD1836_DACL3_MUTE 4 -#define AD1836_DACR3_MUTE 5 - -#define AD1836_DAC_L1_VOL 2 -#define AD1836_DAC_R1_VOL 3 -#define AD1836_DAC_L2_VOL 4 -#define AD1836_DAC_R2_VOL 5 -#define AD1836_DAC_L3_VOL 6 -#define AD1836_DAC_R3_VOL 7 - -#define AD1836_ADC_CTRL1 12 -#define AD1836_ADC_POWERDOWN 7 -#define AD1836_ADC_HIGHPASS_FILTER 8 - -#define AD1836_ADC_CTRL2 13 -#define AD1836_ADCL1_MUTE 0 -#define AD1836_ADCR1_MUTE 1 -#define AD1836_ADCL2_MUTE 2 -#define AD1836_ADCR2_MUTE 3 -#define AD1836_ADC_WORD_LEN_MASK 0x30 -#define AD1836_ADC_SERFMT_MASK (7 << 6) -#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) -#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) -#define AD1836_ADC_AUX (0x6 << 6) - -#define AD1836_ADC_CTRL3 14 - -#define AD1836_NUM_REGS 16 - -#endif diff --git a/sound/soc/codecs/ad183x.c b/sound/soc/codecs/ad183x.c new file mode 100644 index 0000000..2dd548a --- /dev/null +++ b/sound/soc/codecs/ad183x.c @@ -0,0 +1,474 @@ +/* + * AD183X Audio Codec driver supporting AD1835A, AD1836, AD1837A, AD1838A, AD1839A + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/spi/spi.h> +#include "ad183x.h" + +/* codec private data */ +struct ad183x_chl_ctrls { + struct snd_kcontrol_new *snd_ctrls; + struct snd_soc_dapm_widget *dapm_widgets; + struct snd_soc_dapm_route *audio_paths; + int ctrl_num; + int widget_num; + int path_num; +}; + +struct ad183x_priv { + enum snd_soc_control_type control_type; + struct ad183x_chl_ctrls chl_ctrl; +}; + +/* + * AD183X volume/mute/de-emphasis etc. controls + */ +static const char *ad183x_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; + +static const struct soc_enum ad183x_deemp_enum = +SOC_ENUM_SINGLE(AD183X_DAC_CTRL1, 8, 4, ad183x_deemp); + +/* AD1835A/AD1837A: 4 stereo DAC, 1 stereo ADC; */ +static const struct snd_kcontrol_new ad1835a_ad1837a_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD183X_DAC_L1_VOL, + AD183X_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD183X_DAC_L2_VOL, + AD183X_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD183X_DAC_L3_VOL, + AD183X_DAC_R3_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC4 Volume", AD183X_DAC_L4_VOL, + AD183X_DAC_R4_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD183X_ADC_CTRL2, AD183X_ADCL1_MUTE, + AD183X_ADCR1_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD183X_DAC_CTRL2, AD183X_DACL1_MUTE, + AD183X_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD183X_DAC_CTRL2, AD183X_DACL2_MUTE, + AD183X_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD183X_DAC_CTRL2, AD183X_DACL3_MUTE, + AD183X_DACR3_MUTE, 1, 1), + SOC_DOUBLE("DAC4 Switch", AD183X_DAC_CTRL2, AD183X_DACL4_MUTE, + AD183X_DACR4_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD183X_ADC_CTRL1, + AD183X_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad183x_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1835a_ad1837a_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD183X_DAC_CTRL1, + AD183X_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD183X_ADC_CTRL1, + AD183X_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), +}; + +static const struct snd_soc_dapm_route ad1835a_ad1837a_audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "DAC3OUT", "DAC4 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, +}; + +/* AD1836: 3 stereo DAC, 2 stereo ADC; */ +static const struct snd_kcontrol_new ad1836_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD183X_DAC_L1_VOL, + AD183X_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD183X_DAC_L2_VOL, + AD183X_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD183X_DAC_L3_VOL, + AD183X_DAC_R3_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD183X_ADC_CTRL2, AD183X_ADCL1_MUTE, + AD183X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD183X_ADC_CTRL2, AD183X_ADCL2_MUTE, + AD183X_ADCR2_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD183X_DAC_CTRL2, AD183X_DACL1_MUTE, + AD183X_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD183X_DAC_CTRL2, AD183X_DACL2_MUTE, + AD183X_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD183X_DAC_CTRL2, AD183X_DACL3_MUTE, + AD183X_DACR3_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD183X_ADC_CTRL1, + AD183X_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad183x_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD183X_DAC_CTRL1, + AD183X_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD183X_ADC_CTRL1, + AD183X_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), +}; + +static const struct snd_soc_dapm_route ad1836_audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, +}; + +/* AD1838A/AD1939A: 3 stereo DAC, 1 stereo ADC; */ +static const struct snd_kcontrol_new ad1838a_ad1839a_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD183X_DAC_L1_VOL, + AD183X_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD183X_DAC_L2_VOL, + AD183X_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD183X_DAC_L3_VOL, + AD183X_DAC_R3_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD183X_ADC_CTRL2, AD183X_ADCL1_MUTE, + AD183X_ADCR1_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD183X_DAC_CTRL2, AD183X_DACL1_MUTE, + AD183X_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD183X_DAC_CTRL2, AD183X_DACL2_MUTE, + AD183X_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD183X_DAC_CTRL2, AD183X_DACL3_MUTE, + AD183X_DACR3_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD183X_ADC_CTRL1, + AD183X_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad183x_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1838a_ad1839a_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD183X_DAC_CTRL1, + AD183X_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD183X_ADC_CTRL1, + AD183X_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), +}; + +static const struct snd_soc_dapm_route ad1838a_ad1839a_audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, +}; + +/* + * DAI ops entries + */ + +static int ad183x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + /* at present, we support adc aux mode to interface with + * blackfin sport tdm mode + */ + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + /* ALCLK,ABCLK are both output, AD183X can only be master */ + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ad183x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int word_len = 0; + + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = 3; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_len = 1; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + word_len = 0; + break; + } + + snd_soc_update_bits(codec, AD183X_DAC_CTRL1, + AD183X_DAC_WORD_LEN_MASK, word_len); + + snd_soc_update_bits(codec, AD183X_ADC_CTRL2, + AD183X_ADC_WORD_LEN_MASK, word_len); + return 0; +} + +static struct snd_soc_dai_ops ad183x_dai_ops = { + .hw_params = ad183x_hw_params, + .set_fmt = ad183x_set_dai_fmt, +}; + +/* codec DAI instance */ +static struct snd_soc_dai_driver ad183x_dai = { + .name = "ad183x-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad183x_dai_ops, +}; + +static int ad183x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + /* reset clock control mode */ + u16 adc_ctrl2 = snd_soc_read(codec, AD183X_ADC_CTRL2); + adc_ctrl2 &= ~AD183X_ADC_SERFMT_MASK; + + return snd_soc_write(codec, AD183X_ADC_CTRL2, adc_ctrl2); +} + +static int ad183x_resume(struct snd_soc_codec *codec) +{ + /* restore clock control mode */ + u16 adc_ctrl2 = snd_soc_read(codec, AD183X_ADC_CTRL2); + adc_ctrl2 |= AD183X_ADC_AUX; + + return snd_soc_write(codec, AD183X_ADC_CTRL2, adc_ctrl2); +} + +static int ad183x_probe(struct snd_soc_codec *codec) +{ + struct ad183x_priv *ad183x = snd_soc_codec_get_drvdata(codec); + struct ad183x_chl_ctrls *chl_ctrl; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 4, 12, ad183x->control_type); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + /* default setting for ad183x */ + /* de-emphasis: 48kHz, power-on dac */ + snd_soc_write(codec, AD183X_DAC_CTRL1, 0x300); + /* unmute dac channels */ + snd_soc_write(codec, AD183X_DAC_CTRL2, 0x0); + /* high-pass filter enable, power-on adc */ + snd_soc_write(codec, AD183X_ADC_CTRL1, 0x100); + /* unmute adc channles, adc aux mode */ + snd_soc_write(codec, AD183X_ADC_CTRL2, 0x180); + /* left/right diff:PGA/MUX */ + snd_soc_write(codec, AD183X_ADC_CTRL3, 0x3A); + /* volume */ + snd_soc_write(codec, AD183X_DAC_L1_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_R1_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_L2_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_R2_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_L3_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_R3_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_L4_VOL, 0x3FF); + snd_soc_write(codec, AD183X_DAC_R4_VOL, 0x3FF); + + chl_ctrl = &ad183x->chl_ctrl; + + snd_soc_add_controls(codec, chl_ctrl->snd_ctrls, chl_ctrl->ctrl_num); + snd_soc_dapm_new_controls(dapm, chl_ctrl->dapm_widgets, chl_ctrl->widget_num); + snd_soc_dapm_add_routes(dapm, chl_ctrl->audio_paths, chl_ctrl->path_num); + + return 0; +} + +/* power down chip */ +static int ad183x_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ad183x = { + .probe = ad183x_probe, + .remove = ad183x_remove, + .suspend = ad183x_suspend, + .resume = ad183x_resume, + .reg_cache_size = AD183X_NUM_REGS, + .reg_word_size = sizeof(u16), +}; + +static int __devinit ad183x_spi_probe(struct spi_device *spi) +{ + struct ad183x_priv *ad183x; + int ret; + char *chip_name = spi->dev.platform_data; + + if (!chip_name) + return -ENODEV; + + ad183x = kzalloc(sizeof(struct ad183x_priv), GFP_KERNEL); + if (ad183x == NULL) + return -ENOMEM; + + if ((strcmp(chip_name, "ad1835a") == 0) || (strcmp(chip_name, "ad1837a") == 0)) { + ad183x->chl_ctrl.snd_ctrls = (struct snd_kcontrol_new *)ad1835a_ad1837a_snd_controls; + ad183x->chl_ctrl.dapm_widgets = (struct snd_soc_dapm_widget *)ad1835a_ad1837a_dapm_widgets; + ad183x->chl_ctrl.audio_paths = (struct snd_soc_dapm_route *)ad1835a_ad1837a_audio_paths; + ad183x->chl_ctrl.ctrl_num = ARRAY_SIZE(ad1835a_ad1837a_snd_controls); + ad183x->chl_ctrl.widget_num = ARRAY_SIZE(ad1835a_ad1837a_dapm_widgets); + ad183x->chl_ctrl.path_num = ARRAY_SIZE(ad1835a_ad1837a_audio_paths); + ad183x_dai.playback.channels_max = 8; + ad183x_dai.capture.channels_max = 2; + } else if ((strcmp(chip_name, "ad1838a") == 0) || (strcmp(chip_name, "ad1839a") == 0)) { + ad183x->chl_ctrl.snd_ctrls = (struct snd_kcontrol_new *)ad1838a_ad1839a_snd_controls; + ad183x->chl_ctrl.dapm_widgets = (struct snd_soc_dapm_widget *)ad1838a_ad1839a_dapm_widgets; + ad183x->chl_ctrl.audio_paths = (struct snd_soc_dapm_route *)ad1838a_ad1839a_audio_paths; + ad183x->chl_ctrl.ctrl_num = ARRAY_SIZE(ad1838a_ad1839a_snd_controls); + ad183x->chl_ctrl.widget_num = ARRAY_SIZE(ad1838a_ad1839a_dapm_widgets); + ad183x->chl_ctrl.path_num = ARRAY_SIZE(ad1838a_ad1839a_audio_paths); + ad183x_dai.playback.channels_max = 6; + ad183x_dai.capture.channels_max = 2; + } else if (strcmp(chip_name, "ad1836") == 0) { + ad183x->chl_ctrl.snd_ctrls = (struct snd_kcontrol_new *)ad1836_snd_controls; + ad183x->chl_ctrl.dapm_widgets = (struct snd_soc_dapm_widget *)ad1836_dapm_widgets; + ad183x->chl_ctrl.audio_paths = (struct snd_soc_dapm_route *)ad1836_audio_paths; + ad183x->chl_ctrl.ctrl_num = ARRAY_SIZE(ad1836_snd_controls); + ad183x->chl_ctrl.widget_num = ARRAY_SIZE(ad1836_dapm_widgets); + ad183x->chl_ctrl.path_num = ARRAY_SIZE(ad1836_audio_paths); + ad183x_dai.playback.channels_max = 6; + ad183x_dai.capture.channels_max = 4; + } else { + dev_err(&spi->dev, "not supported chip type\n"); + return -EINVAL; + } + + spi_set_drvdata(spi, ad183x); + ad183x->control_type = SND_SOC_SPI; + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_ad183x, &ad183x_dai, 1); + + if (ret < 0) + kfree(ad183x); + return ret; +} + +static int __devexit ad183x_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + + return 0; +} + +static struct spi_driver ad183x_spi_driver = { + .driver = { + .name = "ad183x-codec", + .owner = THIS_MODULE, + }, + .probe = ad183x_spi_probe, + .remove = __devexit_p(ad183x_spi_remove), +}; + +static int __init ad183x_init(void) +{ + int ret = 0; +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&ad183x_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register ad183x SPI driver: %d\n", + ret); + } +#endif + return ret; +} +module_init(ad183x_init); + +static void __exit ad183x_exit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&ad183x_spi_driver); +#endif +} +module_exit(ad183x_exit); + +MODULE_DESCRIPTION("ASoC ad183x driver"); +MODULE_AUTHOR("Barry Song <21cnbao(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad183x.h b/sound/soc/codecs/ad183x.h new file mode 100644 index 0000000..f7af3db --- /dev/null +++ b/sound/soc/codecs/ad183x.h @@ -0,0 +1,57 @@ +/* + * AD183X Audio Codec driver supporting AD1835A, AD1836, AD1837A, AD1838A, AD1839A + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __AD183X_H__ +#define __AD183X_H__ + +#define AD183X_DAC_CTRL1 0 +#define AD183X_DAC_POWERDOWN 2 +#define AD183X_DAC_SERFMT_MASK 0xE0 +#define AD183X_DAC_SERFMT_PCK256 (0x4 << 5) +#define AD183X_DAC_SERFMT_PCK128 (0x5 << 5) +#define AD183X_DAC_WORD_LEN_MASK 0x18 + +#define AD183X_DAC_CTRL2 1 +#define AD183X_DACL1_MUTE 0 +#define AD183X_DACR1_MUTE 1 +#define AD183X_DACL2_MUTE 2 +#define AD183X_DACR2_MUTE 3 +#define AD183X_DACL3_MUTE 4 +#define AD183X_DACR3_MUTE 5 +#define AD183X_DACL4_MUTE 6 +#define AD183X_DACR4_MUTE 7 + +#define AD183X_DAC_L1_VOL 2 +#define AD183X_DAC_R1_VOL 3 +#define AD183X_DAC_L2_VOL 4 +#define AD183X_DAC_R2_VOL 5 +#define AD183X_DAC_L3_VOL 6 +#define AD183X_DAC_R3_VOL 7 +#define AD183X_DAC_L4_VOL 8 +#define AD183X_DAC_R4_VOL 9 + +#define AD183X_ADC_CTRL1 12 +#define AD183X_ADC_POWERDOWN 7 +#define AD183X_ADC_HIGHPASS_FILTER 8 + +#define AD183X_ADC_CTRL2 13 +#define AD183X_ADCL1_MUTE 0 +#define AD183X_ADCR1_MUTE 1 +#define AD183X_ADCL2_MUTE 2 +#define AD183X_ADCR2_MUTE 3 +#define AD183X_ADC_WORD_LEN_MASK 0x30 +#define AD183X_ADC_SERFMT_MASK (7 << 6) +#define AD183X_ADC_SERFMT_PCK256 (0x4 << 6) +#define AD183X_ADC_SERFMT_PCK128 (0x5 << 6) +#define AD183X_ADC_AUX (0x6 << 6) + +#define AD183X_ADC_CTRL3 14 + +#define AD183X_NUM_REGS 16 + +#endif -- 1.7.4.1

2 1

Re: [alsa-devel] Loud pops from Intel HDA onboard sound, may damage speakers (fwd)
by Wu Fengguang 26 Mar '11

26 Mar '11

Hi Chris, Would you run http://www.alsa-project.org/alsa-info.sh and email the results? Thanks, Fengguang On Thu, Mar 24, 2011 at 01:35:54AM +0000, Chris Wilson wrote: > ---------- Forwarded message ---------- > Date: Thu, 24 Mar 2011 00:49:55 +0000 (GMT) > From: Chris Wilson <chris(a)qwirx.com> > To: Takashi Iwai <tiwai(a)suse.de>, PeiSen Hou <pshou(a)realtek.com.tw>, > matt.jared(a)intel.com, andy.kopp(a)intel.com, dan.d.kogan(a)intel.com > Subject: Loud pops from Intel HDA onboard sound, may damage speakers > > Dear sirs, > > I'm writing to you as the named contacts in hda_intel.c in the Linux > kernel. Please let me know if I should contact someone else instead. > > I refer to this downstream bug on Ubuntu Launchpad, which I am > hereby taking upstream: > https://bugs.launchpad.net/ubuntu/+source/linux/+bug/184632 > > I have a desktop system (Fujitsu P9900) with the onboard sound > connected to my hi-fi amplifier, driven by snd_hda_intel. The audio > chipset is: > > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset > High Definition Audio (rev 05) > > The sound card outputs loud popping sounds whenever I suspend and > resume it. One of them is loud or severe enough to cause my hi-fi > amplifier to go into automatic shutdown mode, and I'm worried that > it may damage my speakers. > > If I enable power_save=1, as suggested by powertop, then this > happens about 2 seconds after any audio finishes playing as well, so > I suspect it's related to putting the sound card in or out of > suspend mode. > > The pops seem to be happening to others who have this chipset as > well, as seen on the bug report, but they seem mostly to be using > laptops or otherwise not bothered by potential damage to their > speakers. > > I would really appreciate it if you could help me to find a way to > avoid these pops, for example by muting the audio output before > shutting down the DAC? I think it would be great if this could make > it into the standard kernel driver. > > Many thanks in advance, > > Chris Wilson. > -- > _____ __ _ > \ __/ / ,__(_)_ | Chris Wilson <chris+sig(a)qwirx.com> Cambs UK | > / (_/ ,\/ _/ /_ \ | Security/C/C++/Java/Ruby/Perl/SQL Developer | > \__/_/_/_//_/___/ | We are GNU : free your mind & your software |

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[alsa-devel] [PATCH 1/3] Support single-rate no-SRC cards.
by linux＠audioscience.com 25 Mar '11

25 Mar '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Cards without settable local samplerate and without SRC still must have a valid samplerate. This fixed rate is determined by reading the current rate for the card. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- pci/asihpi/asihpi.c | 16 +++++++--------- 1 files changed, 7 insertions(+), 9 deletions(-) diff --git a/pci/asihpi/asihpi.c b/pci/asihpi/asihpi.c index 0ac1f98..22606e3 100644 --- a/pci/asihpi/asihpi.c +++ b/pci/asihpi/asihpi.c @@ -381,13 +381,13 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, "No local sampleclock, err %d\n", err); } - for (idx = 0; idx < 100; idx++) { - if (hpi_sample_clock_query_local_rate( - h_control, idx, &sample_rate)) { - if (!idx) - snd_printk(KERN_ERR - "Local rate query failed\n"); - + for (idx = -1; idx < 100; idx++) { + if (idx == -1) { + if (hpi_sample_clock_get_sample_rate(h_control, + &sample_rate)) + continue; + } else if (hpi_sample_clock_query_local_rate(h_control, + idx, &sample_rate)) { break; } @@ -440,8 +440,6 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, } } - /* printk(KERN_INFO "Supported rates %X %d %d\n", - rates, rate_min, rate_max); */ pcmhw->rates = rates; pcmhw->rate_min = rate_min; pcmhw->rate_max = rate_max; -- 1.7.0.4

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[alsa-devel] snd_soc_dapm_force_enable_pin question
by Koul, Vinod 25 Mar '11

25 Mar '11

Hi, I have a Vibra controller on the codec which can be programmed using alsa controls. Now after configuration I added an enable control which when turned On, forces the Vibra Pin to be ON. The Pin is on, but not the path, I need the DAC and amplifier connected to be ON as well? Is this not what the force_enable_pin() is supposed to do? If not how should I ensure this path and codec BIAS is on when Vibra is activated by user? -- ~Vinod

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[alsa-devel] QasMixer 0.10.1 available
by Sebastian H. 25 Mar '11

25 Mar '11

Hello Alsa and Qt users QasMixer version 0.10.1 is now available. This release fixes some issues with session mangement and the spanish translation is updated. Changes: * Improved single instance check * Fix for not shutting down on a session shutdown * Spanish translation updated (Jose L.) Homepage with more informations and screenshots http://xwmw.org/qasmixer Files are hosted at sourceforge http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

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[alsa-devel] no external mic
by leon 25 Mar '11

25 Mar '11

Hello, i've Laptop Dell Inspiron 1520, the internal mic works perfectly, but external mic doesn't work, i've followed this link: http://ubuntuforums.org/showthread.php?p=10352564 but it didn't help me, here's the output of ALSA Information Script: http://www.alsa-project.org/db/?f=3aa8a44412081f533b2de3e32aae76ff4370eae9 btw, i've been using Ubuntu, and Debian now, both have same problem. so i haven't tried other Distributions. any idea? thanks. ----------------------------------------------------------------------------------------------------------------------- Send big files for free. Simple steps. No registration. Visit now http://www.nawelny.com

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[alsa-devel] DAPM widgets beyond 2.6.38
by Peter Hsiang 25 Mar '11

25 Mar '11

1) In 2.6.38, after these 2 operations: snd_soc_dapm_new_controls(dapm, dapm_widgets, ARRAY_SIZE(dapm_widgets)); snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); Is it *always* necessary to call snd_soc_dapm_new_widgets(dapm)? I have seen one case where it was not called. 2) In 2.6.39, to do the same, is there any function to call other than the table entries? Thanks, Peter

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