December 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Implement WM8903 high pass filter support
by Mark Brown 11 Dec '10

11 Dec '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 620793e..797992c 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -474,6 +474,13 @@ static const DECLARE_TLV_DB_SCALE(drc_tlv_min, 0, 600, 0); static const DECLARE_TLV_DB_SCALE(drc_tlv_max, 1200, 600, 0); static const DECLARE_TLV_DB_SCALE(drc_tlv_startup, -300, 50, 0); +static const char *hpf_mode_text[] = { + "Hi-fi", "Voice 1", "Voice 2", "Voice 3" +}; + +static const struct soc_enum hpf_mode = + SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text); + static const char *drc_slope_text[] = { "1", "1/2", "1/4", "1/8", "1/16", "0" }; @@ -612,6 +619,8 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, 6, 1, 0), /* ADCs */ +SOC_SINGLE("HPF Switch", WM8903_ADC_DIGITAL_0, 4, 1, 0), +SOC_ENUM("HPF Mode", hpf_mode), SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0), SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0), SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1), -- 1.7.1

2 1

[alsa-devel] [PATCH v2] ASoC: tlv320dac33: Power down digital parts, when not needed
by Peter Ujfalusi 10 Dec '10

10 Dec '10

If the following scenario has been followed: 1. Enable analog bypass amixer sset 'Analog Left Bypass' on amixer sset 'Analog Right Bypass' on 2. Start playback aplay -fdat -d3 /dev/zero After the playback stopped (3 sec), and the soc timeout (5 sec), the digital parts of the codec will remain powered up. This means that the DAI clocks are continue to run, the oscillator remain operational, etc. Use the SND_SOC_DAPM_POST_PMD widget to get notification about the stopped stream, and power down the digital part of the codec. If the analog bypass is enabled, than the codec will remain in BIAS_ON level, and things will work correctly. In case, if the bypass is disabled, than the codec will fall to BIAS_STANDBY than to BIAS_OFF level, as it used to. The digital part of DAC33 is initialized at every stream start (DAPM_PRE:PRE_PMU event), so subsequent streams (within 5 sec) will have working DAI. When the codec is coming out from BIAS_OFF, the full power-up sequence followed by the same DAPM_PRE widget event will power up the digital part. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320dac33.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index b3445b3..776ac80 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -354,6 +354,21 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } +static inline void dac33_disable_digital(struct snd_soc_codec *codec) +{ + u8 reg; + + /* Stop the DAI clock */ + reg = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + reg &= ~DAC33_BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg); + + /* Power down the Oscillator, and DACs */ + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + reg &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); @@ -402,7 +417,7 @@ exit: return ret; } -static int playback_event(struct snd_soc_dapm_widget *w, +static int dac33_playback_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec); @@ -414,6 +429,9 @@ static int playback_event(struct snd_soc_dapm_widget *w, dac33_prepare_chip(dac33->substream); } break; + case SND_SOC_DAPM_POST_PMD: + dac33_disable_digital(w->codec); + break; } return 0; } @@ -609,7 +627,8 @@ 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_PRE("Prepare Playback", playback_event), + SND_SOC_DAPM_PRE("Pre Playback", dac33_playback_event), + SND_SOC_DAPM_POST("Post Playback", dac33_playback_event), }; static const struct snd_soc_dapm_route audio_map[] = { -- 1.7.3.3

2 1

[alsa-devel] [PATCH v4] ASoC: Add HP iPAQ H1940 support
by Vasily Khoruzhick 10 Dec '10

10 Dec '10

Add glue driver to make s3c24xx-i2s and uda1380 produce some sound on H1940. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v4: Adapt v3 version to latest -next tree v3-v1: History lost :( sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/h1940_uda1380.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/h1940_uda1380.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index eb45cf9..67cdad4 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -116,6 +116,14 @@ config ASOC_SAMSUNG_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select ASOC_SAMSUNG_SIMTEC +config ASOC_SAMSUNG_H1940_UDA1380 + tristate "Audio support for the HP iPAQ H1940" + depends on ASOC_SAMSUNG && ARCH_H1940 + select SND_S3C24XX_I2S + select SND_SOC_UDA1380 + help + This driver provides audio support for HP iPAQ h1940 PDA. + config ASOC_SAMSUNG_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on ASOC_SAMSUNG && MACH_RX1950 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 0d24f95..622e76e 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -27,6 +27,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o snd-soc-smdk-wm8580-objs := smdk_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o @@ -43,6 +44,7 @@ obj-$(CONFIG_ASOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_ASOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_ASOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o obj-$(CONFIG_ASOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o obj-$(CONFIG_ASOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c new file mode 100644 index 0000000..c45f7ce --- /dev/null +++ b/sound/soc/samsung/h1940_uda1380.c @@ -0,0 +1,296 @@ +/* + * h1940-uda1380.c -- ALSA Soc Audio Layer + * + * Copyright (c) 2010 Arnaud Patard <arnaud.patard(a)rtp-net.org> + * Copyright (c) 2010 Vasily Khoruzhick <anarsoul(a)gmail.com> + * + * Based on version from Arnaud Patard <arnaud.patard(a)rtp-net.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/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> + +#include <sound/soc.h> +#include <sound/uda1380.h> +#include <sound/jack.h> + +#include <plat/regs-iis.h> + +#include <mach/h1940-latch.h> + +#include <asm/mach-types.h> + +#include "dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h" + +static unsigned int rates[] = { + 11025, + 22050, + 44100, +}; + +static struct snd_pcm_hw_constraint_list hw_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +static struct snd_soc_jack_gpio hp_jack_gpios[] = { + { + .gpio = S3C2410_GPG(4), + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .invert = 1, + .debounce_time = 200, + }, +}; + +static int h1940_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = hw_rates.list[0]; + runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_rates); +} + +static int h1940_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_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int div; + int ret; + unsigned int rate = params_rate(params); + + switch (rate) { + case 11025: + case 22050: + case 44100: + div = s3c24xx_i2s_get_clockrate() / (384 * rate); + if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) + div++; + break; + default: + dev_err(&rtd->dev, "%s: rate %d is not supported\n", + __func__, rate); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* select clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + /* set BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops h1940_ops = { + .startup = h1940_startup, + .hw_params = h1940_hw_params, +}; + +static int h1940_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(H1940_LATCH_AUDIO_POWER, 1); + else + gpio_set_value(H1940_LATCH_AUDIO_POWER, 0); + + return 0; +} + +/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), +}; + +/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to VOUTLHP, VOUTRHP */ + {"Headphone Jack", NULL, "VOUTLHP"}, + {"Headphone Jack", NULL, "VOUTRHP"}, + + /* ext speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* mic is connected to VINM */ + {"VINM", NULL, "Mic Jack"}, +}; + +static struct platform_device *s3c24xx_snd_device; + +static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int err; + + /* Add h1940 specific widgets */ + err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + if (err) + return err; + + /* Set up h1940 specific audio path audio_mapnects */ + err = snd_soc_dapm_add_routes(dapm, audio_map, + ARRAY_SIZE(audio_map)); + if (err) + return err; + + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + + snd_soc_dapm_sync(dapm); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack); + + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + + snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + +/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = { + { + .name = "uda1380", + .stream_name = "UDA1380 Duplex", + .cpu_dai_name = "s3c24xx-iis", + .codec_dai_name = "uda1380-hifi", + .init = h1940_uda1380_init, + .platform_name = "samsung-audio", + .codec_name = "uda1380-codec.0-001a", + .ops = &h1940_ops, + }, +}; + +static struct snd_soc_card h1940_asoc = { + .name = "h1940", + .dai_link = h1940_uda1380_dai, + .num_links = ARRAY_SIZE(h1940_uda1380_dai), +}; + +static int __init h1940_init(void) +{ + int ret; + + if (!machine_is_h1940()) + return -ENODEV; + + /* configure some gpios */ + ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power"); + if (ret) + goto err_out; + + ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0); + if (ret) + goto err_gpio; + + s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_snd_device) { + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc); + ret = platform_device_add(s3c24xx_snd_device); + + if (ret) + goto err_plat; + + return 0; + +err_plat: + platform_device_put(s3c24xx_snd_device); +err_gpio: + gpio_free(H1940_LATCH_AUDIO_POWER); + +err_out: + return ret; +} + +static void __exit h1940_exit(void) +{ + platform_device_unregister(s3c24xx_snd_device); + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + gpio_free(H1940_LATCH_AUDIO_POWER); +} + +module_init(h1940_init); +module_exit(h1940_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL"); -- 1.7.3.2

3 2

[alsa-devel] [PATCH] ASoC: Implement WM8994/58 DAC and ADC oversampling control
by Mark Brown 10 Dec '10

10 Dec '10

The oversampling rate of the DAC and ADC can be controlled to optimise for either low power consumption or maximum performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b599616..af104ac 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -558,6 +558,16 @@ static const struct soc_enum aif2dacl_src = static const struct soc_enum aif2dacr_src = SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text); +static const char *osr_text[] = { + "Low Power", "High Performance", +}; + +static const struct soc_enum dac_osr = + SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text); + +static const struct soc_enum adc_osr = + SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text); + static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -834,6 +844,9 @@ SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0), SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf), SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0), +SOC_ENUM("ADC OSR", adc_osr), +SOC_ENUM("DAC OSR", dac_osr), + SOC_DOUBLE_R_TLV("DAC1 Volume", WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), SOC_DOUBLE_R("DAC1 Switch", WM8994_DAC1_LEFT_VOLUME, -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: Optimise WM9081 FLL performance
by Mark Brown 10 Dec '10

10 Dec '10

From: Mario Becroft <mb(a)gem.win.co.nz> Tune the FLL gain for optimal performance according to evaluation results. Signed-off-by: Mario Becroft <mb(a)gem.win.co.nz> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index c706077..e5055b2 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -590,6 +590,10 @@ static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id, reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT; snd_soc_write(codec, WM9081_FLL_CONTROL_5, reg5); + /* Set gain to the recommended value */ + snd_soc_update_bits(codec, WM9081_FLL_CONTROL_4, + WM9081_FLL_GAIN_MASK, 0); + /* Enable the FLL */ snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA); -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Power down digital parts, when not needed
by Peter Ujfalusi 10 Dec '10

10 Dec '10

If the following scenarion has been followed: 1. Enable analog bypass amixer sset 'Analog Left Bypass' on amixer sset 'Analog Right Bypass' on 2. Start playback aplay -fdat -d3 /dev/zero After the playback stopped (3 sec), and the soc timeout (5 sec), the digital parts of the codec will remain powered up. This means that the DAI clocks are continue to run, the oscillator remain operational, etc. Use the SND_SOC_DAPM_POST_PMD widget to get notification about the stopped stream, and power down the digital part of the codec. If the analog bypass is enabled, than the codec will remain in BIAS_ON level, and things will work correctly. In case, if the bypass is disabled, than the codec will fall to BIAS_STANDBY than to BIAS_OFF level, as it used to. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index b3445b3..776ac80 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -354,6 +354,21 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) dac33_write(codec, DAC33_PWR_CTRL, reg); } +static inline void dac33_disable_digital(struct snd_soc_codec *codec) +{ + u8 reg; + + /* Stop the DAI clock */ + reg = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + reg &= ~DAC33_BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg); + + /* Power down the Oscillator, and DACs */ + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + reg &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); @@ -402,7 +417,7 @@ exit: return ret; } -static int playback_event(struct snd_soc_dapm_widget *w, +static int dac33_playback_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec); @@ -414,6 +429,9 @@ static int playback_event(struct snd_soc_dapm_widget *w, dac33_prepare_chip(dac33->substream); } break; + case SND_SOC_DAPM_POST_PMD: + dac33_disable_digital(w->codec); + break; } return 0; } @@ -609,7 +627,8 @@ 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_PRE("Prepare Playback", playback_event), + SND_SOC_DAPM_PRE("Pre Playback", dac33_playback_event), + SND_SOC_DAPM_POST("Post Playback", dac33_playback_event), }; static const struct snd_soc_dapm_route audio_map[] = { -- 1.7.3.3

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[alsa-devel] [PATCH] ALSA: aoa: Remove wrong i2c_set_clientdata in onyx_i2c_remove()
by Axel Lin 10 Dec '10

10 Dec '10

It does not make sense to set clientdata to onyx in onyx_i2c_remove() as we are going to kfree onyx. What we really want here is i2c_set_clientdata(client, NULL); Since the i2c core will take care of it now, so this patch just removes it. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/aoa/codecs/onyx.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 91852e4..3687a6c 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1114,7 +1114,6 @@ static int onyx_i2c_remove(struct i2c_client *client) of_node_put(onyx->codec.node); if (onyx->codec_info) kfree(onyx->codec_info); - i2c_set_clientdata(client, onyx); kfree(onyx); return 0; } -- 1.7.2

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Re: [alsa-devel] Help wanted: emachines em350 internal mic
by Raymond Yau 09 Dec '10

09 Dec '10

>> Greetings, >> There is only a single mic on this netbook. However, the alsa device >> shows up as stereo, and the right channel carries an inverted copy of >> the left channel. http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=59c774ed5ee00e9623a2… >> So when the two channels are summed to make a mono (e.g. for >> telephony, or playback on mono internal speaker) it appears that there >> is no signal. >> Is there a way to "cure" this problem in the driver? I can think of >> * Force internal mic capture to be mono (left only) >> * Copy left to right capture >> * Invert right capture >> but don't know how easy any are to implement.

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 09 Dec '10

09 Dec '10

The following changes since commit 146fd574ec06b1c593805738b1c2c8c5a4128681: ASoC: Add ADC high pass filter support to WM8994 (2010-12-08 15:46:49 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Alexander Sverdlin (1): ASoC: EP93xx: sampling rate range extended Dmitry Artamonow (1): ASoC: fix deemphasis control in wm8904/55/60 codecs Mark Brown (1): Merge branch 'for-2.6.37' into for-2.6.38 Seungwhan Youn (2): ASoC: WM8580: Fix R8 initial value ASoC: Remove unnecessary structure definitions include/sound/soc.h | 3 --- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8904.c | 3 ++- sound/soc/codecs/wm8955.c | 3 ++- sound/soc/codecs/wm8960.c | 3 ++- sound/soc/ep93xx/ep93xx-i2s.c | 4 ++-- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 7 files changed, 11 insertions(+), 11 deletions(-)

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[alsa-devel] ASoC updates for 2.6.37
by Mark Brown 09 Dec '10

09 Dec '10

The following changes since commit 2a7b1a00206895cfa444fd83477dca67a88a9d25: ASoC: Correct WM8962 interrupt mask register read (2010-12-07 15:49:42 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.37 The usual small, focused bug fixes. Dmitry Artamonow (1): ASoC: fix deemphasis control in wm8904/55/60 codecs Seungwhan Youn (1): ASoC: WM8580: Fix R8 initial value sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8904.c | 3 ++- sound/soc/codecs/wm8955.c | 3 ++- sound/soc/codecs/wm8960.c | 3 ++- 4 files changed, 7 insertions(+), 4 deletions(-)

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