June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 2/4] ASoC: add dialog da9034 (aka micco) audio codec support
by Eric Miao 03 Jun '09

03 Jun '09

Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Bin Yang <bin.yang(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9034-audio.c | 696 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 702 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/da9034-audio.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 121d63f..d779f40 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_DA9034 if PMIC_DA903X help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -160,3 +161,6 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + +config SND_SOC_DA9034 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d8e15a4..53bb3d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,7 @@ snd-soc-wm8990-objs := wm8990.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +snd-soc-da9034-objs := da9034-audio.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o @@ -64,3 +65,4 @@ obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o +obj-$(CONFIG_SND_SOC_DA9034) += snd-soc-da9034.o diff --git a/sound/soc/codecs/da9034-audio.c b/sound/soc/codecs/da9034-audio.c new file mode 100644 index 0000000..159a883 --- /dev/null +++ b/sound/soc/codecs/da9034-audio.c @@ -0,0 +1,696 @@ +/* + * linux/sound/soc/codecs/micco.c + * + * Copyright (C) 2007-2009 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/mfd/da903x.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#define DA9034_MUX_MONO 0x70 +#define DA9034_MUX_BEAR 0x71 +#define DA9034_MUX_LINE 0x72 +#define DA9034_MUX_STEREO_CH1 0x73 +#define DA9034_MUX_STEREO_CH2 0x74 +#define DA9034_AUDIO_LINE_AMP 0x75 +#define DA9034_STEREO_AMP_CH1 0x76 +#define DA9034_STEREO_AMP_CH2 0x77 +#define DA9034_STEREO_DAC 0x78 +#define DA9034_MONO_VOL 0x79 +#define DA9034_BEAR_VOL 0x7a +#define DA9034_I2S_CONTROL 0x7b +#define DA9034_TX_PGA 0x7c +#define DA9034_MIC_PGA 0x7d +#define DA9034_TX_PGA_MUX 0x7e +#define DA9034_VOICE_ADC 0x7f +#define DA9034_VOICE_DAC 0x80 +#define DA9034_SIDE_TONE 0x81 +#define DA9034_PGA_AUX12 0x82 +#define DA9034_PGA_AUX3 0x83 +#define DA9034_PGA_DACS 0x84 + +/* NOTE: the ES1 revision of the chip has the following two issues + * regarding audio: + * 1. The MONO and BEAR are incorrectly exchanged + * 2. The Stereo DAC has to be enabled for Voice Codec + */ +#define DA9034_ERRATA_ES1 1 + +#define ARRAY_AND_SIZE(x) x, ARRAY_SIZE(x) + +/* initial values for the audio registers */ +static const uint8_t da9034_initregs[] = { + 0x00, 0x00, 0x00, 0x00, 0x00, 0x82, 0x00, 0x40, /* 0x70 ~ 0x77 */ + 0xe0, 0x80, 0x80, 0x40, 0x00, 0x40, 0x00, 0x02, /* 0x78 ~ 0x7f */ + 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x80 ~ 0x84 */ +}; + +/* Mono Out Mixer */ +static const struct snd_kcontrol_new da9034_mixer_mono_out_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_MONO, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_MONO, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_MONO, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_MONO, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_MONO, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_MONO, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_MONO, 6, 1, 0), +}; + +static const struct snd_kcontrol_new da9034_mixer_bear_out_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_MONO, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_MONO, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_MONO, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_MONO, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_MONO, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_MONO, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_MONO, 6, 1, 0), +}; + +/* Line Out Mixer */ +static const struct snd_kcontrol_new da9034_mixer_line_out_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_LINE, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_LINE, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_LINE, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_LINE, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_LINE, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_LINE, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_LINE, 6, 1, 0), + SOC_DAPM_SINGLE("MIC", DA9034_MUX_LINE, 7, 1, 0), +}; + +/* Stereo Out Mixer */ +static const struct snd_kcontrol_new da9034_mixer_stereo_ch1_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_STEREO_CH1, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_STEREO_CH1, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_STEREO_CH1, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_STEREO_CH1, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_STEREO_CH1, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_STEREO_CH1, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_STEREO_CH1, 6, 1, 0), +}; + +static const struct snd_kcontrol_new da9034_mixer_stereo_ch2_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_STEREO_CH2, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_STEREO_CH2, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_STEREO_CH2, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_STEREO_CH2, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_STEREO_CH2, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_STEREO_CH2, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_STEREO_CH2, 6, 1, 0), +}; + + +/* default to differential mode for MIC and AUX2 */ +static const char *mux_tx_pga_texts[] = { + "NONE", "MIC", "AUX1", "AUX2", "AUX3", "DAC1", "DAC2", +}; + +static const unsigned int mux_tx_pga_values[] = { + 0x00, 0x0c, 0x02, 0x11, 0x20, 0x40, 0x80, +}; + +static const char *mux_mic_texts[] = { "MIC1", "MIC2" }; + +static const struct soc_enum da9034_mux_enum[] = { + SOC_VALUE_ENUM_SINGLE(DA9034_TX_PGA_MUX, 0, 0xff, 7, + mux_tx_pga_texts, mux_tx_pga_values), + SOC_ENUM_SINGLE(DA9034_MIC_PGA, 4, 2, mux_mic_texts), +}; + +static const struct snd_kcontrol_new mux_ctrls[] = { + SOC_DAPM_VALUE_ENUM("Route", da9034_mux_enum[0]), + SOC_DAPM_ENUM("Route", da9034_mux_enum[1]), +}; + +static const struct snd_kcontrol_new pga_ctrls[] = { + /* PGAs for input */ + SOC_DAPM_SINGLE("MIC Volume", DA9034_MIC_PGA, 0, 0x07, 0), + SOC_DAPM_SINGLE("AUX1 Volume", DA9034_PGA_AUX12, 0, 0x03, 0), + SOC_DAPM_SINGLE("AUX2 Volume", DA9034_PGA_AUX12, 4, 0x03, 0), + SOC_DAPM_SINGLE("AUX3 Volume", DA9034_PGA_AUX3, 0, 0x03, 0), + SOC_DAPM_SINGLE("TX Volume", DA9034_TX_PGA, 1, 0x0f, 0), + + /* PGAs for output */ + SOC_DAPM_SINGLE("MONO Volume", DA9034_MONO_VOL, 0, 0x3f, 0), + SOC_DAPM_SINGLE("BEAR Volume", DA9034_BEAR_VOL, 0, 0x3f, 0), + SOC_DAPM_SINGLE("LINE_OUT Volume", DA9034_AUDIO_LINE_AMP, 0, 0xf, 0), + SOC_DAPM_SINGLE("STEREO_CH1 Volume", DA9034_STEREO_AMP_CH1, 0, 0x3f, 0), + SOC_DAPM_SINGLE("STEREO_CH2 Volume", DA9034_STEREO_AMP_CH2, 0, 0x3f, 0), +}; + +static const struct snd_soc_dapm_widget da9034_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("AUX1"), + SND_SOC_DAPM_INPUT("AUX2"), + SND_SOC_DAPM_INPUT("AUX3"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AUX2inv"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("BEAR"), + SND_SOC_DAPM_OUTPUT("MONO"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + SND_SOC_DAPM_OUTPUT("STEREO_CH1"), + SND_SOC_DAPM_OUTPUT("STEREO_CH2"), + + /* muxers */ + SND_SOC_DAPM_VALUE_MUX("TX-PGA Mux", SND_SOC_NOPM, 0, 0, &mux_ctrls[0]), + SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mux_ctrls[1]), + + /* mixers */ + SND_SOC_DAPM_MIXER("LINE_OUT Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_line_out_controls, 8), + SND_SOC_DAPM_MIXER("MONO Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_mono_out_controls, 7), + SND_SOC_DAPM_MIXER("BEAR Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_bear_out_controls, 7), + SND_SOC_DAPM_MIXER("STEREO_CH1 Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_stereo_ch1_controls, 7), + SND_SOC_DAPM_MIXER("STEREO_CH2 Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_stereo_ch2_controls, 7), + + /* DACs and ADCs */ +#ifdef DA9034_ERRATA_ES1 + SND_SOC_DAPM_DAC("DAC1", "HiFi Playback CH1", -1, 0, 0), + SND_SOC_DAPM_DAC("DAC2", "HiFi Playback CH2", -1, 0, 0), +#else + SND_SOC_DAPM_DAC("DAC1", "HiFi Playback CH1", DA9034_STEREO_DAC, 7, 0), + SND_SOC_DAPM_DAC("DAC2", "HiFi Playback CH2", DA9034_STEREO_DAC, 7, 0), +#endif + SND_SOC_DAPM_DAC("DAC3", "Voice Playback", DA9034_VOICE_DAC, 3, 0), + SND_SOC_DAPM_ADC("VADC", "Voice Capture", DA9034_VOICE_ADC, 0, 0), + + /* PGAs */ + SND_SOC_DAPM_PGA("MIC PGA", DA9034_MIC_PGA, 3, 0, &pga_ctrls[0], 1), + SND_SOC_DAPM_PGA("AUX1 PGA", DA9034_PGA_AUX12, 3, 0, &pga_ctrls[1], 1), + SND_SOC_DAPM_PGA("AUX2 PGA", DA9034_PGA_AUX12, 7, 0, &pga_ctrls[2], 1), + SND_SOC_DAPM_PGA("AUX3 PGA", DA9034_PGA_AUX3, 3, 0, &pga_ctrls[3], 1), + SND_SOC_DAPM_PGA("TX PGA", SND_SOC_NOPM, 0, 0, &pga_ctrls[4], 1), + + SND_SOC_DAPM_PGA("MONO PGA", DA9034_MONO_VOL, 6, 0, &pga_ctrls[5], 1), + SND_SOC_DAPM_PGA("BEAR PGA", DA9034_BEAR_VOL, 6, 0, &pga_ctrls[6], 1), + SND_SOC_DAPM_PGA("LINE_OUT PGA", DA9034_AUDIO_LINE_AMP, 4, 0, + &pga_ctrls[7], 1), + SND_SOC_DAPM_PGA("STEREO_CH1 PGA", DA9034_STEREO_AMP_CH1, 7, 0, + &pga_ctrls[8], 1), + SND_SOC_DAPM_PGA("STEREO_CH2 PGA", DA9034_STEREO_AMP_CH2, 7, 0, + &pga_ctrls[9], 1), + + /* Mic Bias */ + SND_SOC_DAPM_MICBIAS("MIC Int Bias", DA9034_MIC_PGA, 6, 0), + SND_SOC_DAPM_MICBIAS("MIC Ext Bias", DA9034_MIC_PGA, 7, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* MIC PGA */ + { "MIC PGA", NULL, "MIC1" }, + { "MIC PGA", NULL, "MIC2" }, + + /* MONO Mixer */ + { "MONO Mixer", "AUX1", "AUX1" }, + { "MONO Mixer", "AUX2", "AUX2" }, + { "MONO Mixer", "AUX3", "AUX3" }, + { "MONO Mixer", "DAC1", "DAC1" }, + { "MONO Mixer", "DAC2", "DAC2" }, + { "MONO Mixer", "DAC3", "DAC3" }, + { "MONO Mixer", "AUX2inv", "AUX2inv" }, + + { "MONO PGA", NULL, "MONO Mixer" }, +#ifdef DA9034_ERRATA_ES1 + { "BEAR PGA", NULL, "MONO Mixer" }, +#endif + { "MONO", NULL, "MONO PGA" }, + + /* BEAR Mixer */ + { "BEAR Mixer", "AUX1", "AUX1" }, + { "BEAR Mixer", "AUX2", "AUX2" }, + { "BEAR Mixer", "AUX3", "AUX3" }, + { "BEAR Mixer", "DAC1", "DAC1" }, + { "BEAR Mixer", "DAC2", "DAC2" }, + { "BEAR Mixer", "DAC3", "DAC3" }, + { "BEAR Mixer", "AUX2inv", "AUX2inv" }, + +#ifdef DA9034_ERRATA_ES1 + { "MONO PGA", NULL, "BEAR Mixer" }, +#endif + { "BEAR PGA", NULL, "BEAR Mixer" }, + { "BEAR", NULL, "BEAR PGA" }, + + /* LINE_OUT Mixer */ + { "LINE_OUT Mixer", "AUX1", "AUX1" }, + { "LINE_OUT Mixer", "AUX2", "AUX2" }, + { "LINE_OUT Mixer", "AUX3", "AUX3" }, + { "LINE_OUT Mixer", "DAC1", "DAC1" }, + { "LINE_OUT Mixer", "DAC2", "DAC2" }, + { "LINE_OUT Mixer", "DAC3", "DAC3" }, + { "LINE_OUT Mixer", NULL, "MIC PGA" }, + { "LINE_OUT Mixer", "AUX2inv", "AUX2inv" }, + + { "LINE_OUT PGA", NULL, "LINE_OUT Mixer" }, + { "LINE_OUT", NULL, "LINE_OUT PGA" }, + + /* STEREO_CH1 Mixer */ + { "STEREO_CH1 Mixer", "AUX1", "AUX1" }, + { "STEREO_CH1 Mixer", "AUX2", "AUX2" }, + { "STEREO_CH1 Mixer", "AUX3", "AUX3" }, + { "STEREO_CH1 Mixer", "DAC1", "DAC1" }, + { "STEREO_CH1 Mixer", "DAC2", "DAC2" }, + { "STEREO_CH1 Mixer", "DAC3", "DAC3" }, + { "STEREO_CH1 Mixer", "AUX2inv", "AUX2inv" }, + + { "STEREO_CH1 PGA", NULL, "STEREO_CH1 Mixer" }, + { "STEREO_CH1", NULL, "STEREO_CH1 PGA" }, + + /* STEREO_CH2 Mixer */ + { "STEREO_CH2 Mixer", "AUX1", "AUX1" }, + { "STEREO_CH2 Mixer", "AUX2", "AUX2" }, + { "STEREO_CH2 Mixer", "AUX3", "AUX3" }, + { "STEREO_CH2 Mixer", "DAC1", "DAC1" }, + { "STEREO_CH2 Mixer", "DAC2", "DAC2" }, + { "STEREO_CH2 Mixer", "DAC3", "DAC3" }, + { "STEREO_CH2 Mixer", "AUX2inv", "AUX2inv" }, + + { "STEREO_CH2 PGA", NULL, "STEREO_CH2 Mixer" }, + { "STEREO_CH2", NULL, "STEREO_CH2 PGA" }, + + /* TX-PGA P Mux */ + { "TX-PGA Mux", "MIC", "MIC PGA" }, + { "TX-PGA Mux", "AUX1", "AUX1 PGA" }, + { "TX-PGA Mux", "AUX2", "AUX2 PGA" }, + { "TX-PGA Mux", "AUX3", "AUX3 PGA" }, + { "TX-PGA Mux", "DAC1", "DAC1" }, + { "TX-PGA Mux", "DAC2", "DAC2" }, + + { "VADC", NULL, "TX-PGA Mux" }, +}; + +static int da9034_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(da9034_dapm_widgets)); + snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static unsigned int da9034_audio_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + uint8_t val; + + da903x_read(codec->dev, reg, &val); + return val; +} + +static int da9034_audio_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val) +{ + return da903x_write(codec->dev, reg, val); +} + +static int da9034_reset(struct snd_soc_codec *codec, int try_warm) +{ + int reg; + + for (reg = 0; reg < ARRAY_SIZE(da9034_initregs); reg++) + da903x_write(codec->dev, 0x70 + reg, da9034_initregs[reg]); + +#ifdef DA9034_ERRATA_ES1 + da903x_set_bits(codec->dev, DA9034_STEREO_DAC, 1 << 7); +#endif + return 0; +} + +static int da9034_hifi_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + uint8_t i2s_control; + + da903x_read(codec->dev, DA9034_I2S_CONTROL, &i2s_control); + + i2s_control &= 0x0f; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: break; + case SND_SOC_DAIFMT_CBM_CFM: i2s_control |= 0xa0; break; + case SND_SOC_DAIFMT_CBM_CFS: i2s_control |= 0x20; break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + i2s_control |= 0x10; + break; + default: + return -EINVAL; + } + + return da903x_write(codec->dev, DA9034_I2S_CONTROL, i2s_control); +} + +static int da9034_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + uint8_t i2s_control; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + return -EINVAL; + } + + da903x_read(codec->dev, DA9034_I2S_CONTROL, &i2s_control); + + i2s_control &= 0xf0; + + switch (params_rate(params)) { + case 8000: break; + case 11025: i2s_control |= 0x01; break; + case 12000: i2s_control |= 0x02; break; + case 16000: i2s_control |= 0x03; break; + case 22050: i2s_control |= 0x04; break; + case 24000: i2s_control |= 0x05; break; + case 32000: i2s_control |= 0x06; break; + case 44100: i2s_control |= 0x07; break; + case 48000: i2s_control |= 0x0f; break; + default: + return -EINVAL; + } + + return da903x_write(codec->dev, DA9034_I2S_CONTROL, i2s_control); +} + +static int da9034_voice_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + return 0; +} + +static int da9034_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + uint8_t val; + + da903x_read(codec->dev, DA9034_VOICE_ADC, &val); + + val |= 0x18; + + switch ((params_rate(params))) { + case 8000: break; + case 16000: val |= 0x01 << 3; break; + case 32000: val |= 0x02 << 3; break; + default: + return -EINVAL; + } + + return da903x_write(codec->dev, DA9034_VOICE_ADC, val); +} + +#define DA9034_HIFI_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define DA9034_VOICE_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000) + +#define DA9034_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops da9034_dai_ops_hifi = { + .hw_params = da9034_hifi_hw_params, + .set_fmt = da9034_hifi_set_fmt, +}; + +static struct snd_soc_dai_ops da9034_dai_ops_voice = { + .hw_params = da9034_voice_hw_params, + .set_fmt = da9034_voice_set_fmt, +}; + +struct snd_soc_dai da9034_dais[] = { + [0] = { + .name = "I2S HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DA9034_HIFI_RATES, + .formats = DA9034_PCM_FORMATS, + }, + .ops = &da9034_dai_ops_hifi, + }, + [1] = { + .name = "PCM Voice", + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = DA9034_VOICE_RATES, + .formats = DA9034_PCM_FORMATS, + }, + .capture = { + .stream_name = "Voice Capture", + .channels_min = 1, + .channels_max = 1, + .rates = DA9034_VOICE_RATES, + .formats = DA9034_PCM_FORMATS, + }, + .ops = &da9034_dai_ops_voice, + }, +}; +EXPORT_SYMBOL_GPL(da9034_dais); + +#ifdef CONFIG_DEBUG_FS +static struct { + const char *reg_name; + int reg_addr; +} da9034_regs[] = { + { "MUX_MONO", 0x70 }, + { "MUX_BEAR", 0x71 }, + { "MUX_LINE", 0x72 }, + { "MUX_STEREO_CH1", 0x73 }, + { "MUX_STEREO_CH2", 0x74 }, + { "AUDIO_LINE_AMP", 0x75 }, + { "STEREO_AMP_CH1", 0x76 }, + { "STEREO_AMP_CH2", 0x77 }, + { "STEREO_DAC", 0x78 }, + { "MONO_VOL", 0x79 }, + { "BEAR_VOL", 0x7a }, + { "I2S_CONTROL", 0x7b }, + { "TX_PGA", 0x7c }, + { "MIC_AMP", 0x7d }, + { "TX_PGA_MUX", 0x7e }, + { "VOICE_ADC", 0x7f }, + { "VOICE_DAC", 0x80 }, + { "SIDE_TONE", 0x81 }, + { "PGA_AUXI12", 0x82 }, + { "PGA_AUXI3", 0x83 }, + { "PGA_DACS", 0x84 }, +}; + +static int da9034_debugfs_show(struct seq_file *s, void *data) +{ + struct snd_soc_codec *codec = data; + struct device *dev = codec->dev; + uint8_t v; + int i; + + for (i = 0; i < ARRAY_SIZE(da9034_regs); i++) { + da903x_read(codec->dev, da9034_regs[i].reg_addr, &v); + seq_printf(s, "%20s: 0x%02x\n", da9034_regs[i].reg_name, v); + } + return 0; +} + +static int da9034_debugfs_open(struct inode *inode, struct file *file) +{ + return single_open(file, da9034_debugfs_show, inode->i_private); +} + +static const struct file_operations da9034_debugfs_ops = { + .owner = THIS_MODULE, + .open = da9034_debugfs_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void __init da9034_init_debugfs(struct snd_soc_codec *codec) +{ + debugfs_create_file("da9034", S_IFREG | S_IRUGO, NULL, + codec, da9034_debugfs_ops); +} +#else +static inline void da9034_init_debugfs(struct snd_soc_codec *codec) {} +#endif /* CONFIG_DEBUG_FS */ + +static struct snd_soc_codec *da9034_codec; + +static int da9034_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret = 0; + + BUG_ON(!da9034_codec); + + socdev->card->codec = da9034_codec; + socdev->dev = &pdev->dev; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + da9034_add_widgets(da9034_codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + + return 0; + +reset_err: + snd_soc_free_pcms(socdev); + socdev->card->codec = NULL; + return ret; +} + +static int da9034_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_da9034 = { + .probe = da9034_soc_probe, + .remove = da9034_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_da9034); + +static int __devinit da9034_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(*codec), GFP_KERNEL); + if (codec == NULL) { + dev_err(&pdev->dev, "failed to allocate codec\n"); + return -ENOMEM; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = pdev->dev.parent; + codec->name = "DA9034"; + codec->owner = THIS_MODULE; + codec->read = da9034_audio_read; + codec->write = da9034_audio_write; + codec->dai = da9034_dais; + codec->num_dai = ARRAY_SIZE(da9034_dais); + + ret = da9034_reset(codec, 0); + if (ret < 0) { + dev_err(&pdev->dev, "codec reset error\n"); + goto err_free_codec; + } + + ret = snd_soc_register_codec(codec); + if (ret) + goto err_free_codec; + + ret = snd_soc_register_dais(ARRAY_AND_SIZE(da9034_dais)); + if (ret) + goto err_unregister; + + da9034_init_debugfs(codec); + da9034_codec = codec; + platform_set_drvdata(pdev, codec); + + dev_info(&pdev->dev, "da9034 (aka micco) audio codec registered\n"); + return 0; + +err_unregister: + snd_soc_unregister_codec(codec); +err_free_codec: + kfree(codec); + return ret; +} + +static int __devexit da9034_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_codec *codec = platform_get_drvdata(pdev); + + snd_soc_unregister_dais(ARRAY_AND_SIZE(da9034_dais)); + snd_soc_unregister_codec(codec); + kfree(codec); + return 0; +} + +static struct platform_driver da9034_audio_driver = { + .driver = { + .name = "da9034-audio", + .owner = THIS_MODULE, + }, + .probe = da9034_audio_probe, + .remove = da9034_audio_remove, +}; + +static int da9034_audio_init(void) +{ + return platform_driver_register(&da9034_audio_driver); +} +module_init(da9034_audio_init); + +static void da9034_audio_exit(void) +{ + platform_driver_unregister(&da9034_audio_driver); +} +module_exit(da9034_audio_exit); + +MODULE_DESCRIPTION("ASoC DA9034 Audio Codec Driver"); +MODULE_AUTHOR("Bin Yang <bin.yang(a)marvell.com>" + "Paul Shen <bshen9(a)marvell.com>"); +MODULE_LICENSE("GPL"); -- 1.6.0.4

2 1

[alsa-devel] [PATCH 0/2] USB audio gadget driver
by Mike Frysinger 03 Jun '09

03 Jun '09

This is largely a PoC in case people are looking for this kind of thing and want to build upon something. Just throwing it out there at this point. Bryan Wu (2): USB: audio: add USB audio class definitions USB: gadget: add USB Audio Gadget driver drivers/usb/gadget/Kconfig | 14 + drivers/usb/gadget/Makefile | 2 + drivers/usb/gadget/audio.c | 302 ++++++++++++++++++ drivers/usb/gadget/f_audio.c | 707 ++++++++++++++++++++++++++++++++++++++++++ drivers/usb/gadget/u_audio.c | 319 +++++++++++++++++++ drivers/usb/gadget/u_audio.h | 56 ++++ include/linux/usb/audio.h | 265 ++++++++++++++++- 7 files changed, 1663 insertions(+), 2 deletions(-) create mode 100644 drivers/usb/gadget/audio.c create mode 100644 drivers/usb/gadget/f_audio.c create mode 100644 drivers/usb/gadget/u_audio.c create mode 100644 drivers/usb/gadget/u_audio.h

1 2

[alsa-devel] [PATCH 0/4] add da9034 audio support for Littleton and generalize SSP I2S/LeftJ
by Eric Miao 03 Jun '09

03 Jun '09

Thanks for Paul's good work, and finally, we worked together to figure out the generalized I2S/LeftJ support in current pxa-ssp.c, and added the audio support for Littleton: Here's the summary: Paul Shen (4): ASoC: pxa-ssp: enhance I2S and add Left_J support ASoC: add dialog da9034 (aka micco) audio codec support ASoC: pxa add Marvell Littleton platform audio support [ARM] pxa/littleton: add support for audio (da9034) arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 +- arch/arm/mach-pxa/littleton.c | 20 +- include/linux/mfd/da903x.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9034-audio.c | 696 +++++++++++++++++++++++++++++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/littleton.c | 228 ++++++++++ sound/soc/pxa/pxa-ssp.c | 62 ++-- sound/soc/pxa/pxa-ssp.h | 9 + 11 files changed, 1008 insertions(+), 39 deletions(-) -- Cheers - eric

1 1

[alsa-devel] [PATCH 4/4] [ARM] pxa/littleton: add support for audio (da9034)
by Eric Miao 03 Jun '09

03 Jun '09

Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- arch/arm/mach-pxa/littleton.c | 20 +++++++++++++++++++- include/linux/mfd/da903x.h | 1 + 2 files changed, 20 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-pxa/littleton.c b/arch/arm/mach-pxa/littleton.c index e13f6a8..b26b3cb 100644 --- a/arch/arm/mach-pxa/littleton.c +++ b/arch/arm/mach-pxa/littleton.c @@ -81,6 +81,21 @@ static mfp_cfg_t littleton_mfp_cfg[] __initdata = { GPIO27_SSP2_TXD, GPIO17_GPIO, /* SFRM as chip-select */ + /* SSP3 */ + GPIO91_SSP3_SCLK, + GPIO92_SSP3_FRM, + GPIO93_SSP3_TXD, + GPIO94_SSP3_RXD, + /* SSP3 NETWORK CLK */ + GPIO126_EXT_CLK, + + /* SSP4 */ + GPIO95_SSP4_SCLK, + GPIO96_SSP4_FRM, + GPIO97_SSP4_TXD, + GPIO98_SSP4_RXD, + + /* Debug Ethernet */ GPIO90_GPIO, @@ -350,7 +365,10 @@ static struct da903x_subdev_info littleton_da9034_subdevs[] = { }, { .name = "da903x-backlight", .id = DA9034_ID_WLED, - }, + }, { + .name = "da9034-audio", + .id = DA9034_ID_AUDIO, + } }; static struct da903x_platform_data littleton_da9034_info = { diff --git a/include/linux/mfd/da903x.h b/include/linux/mfd/da903x.h index 115dbe9..ba645ff 100644 --- a/include/linux/mfd/da903x.h +++ b/include/linux/mfd/da903x.h @@ -39,6 +39,7 @@ enum { DA9034_ID_VIBRA, DA9034_ID_WLED, DA9034_ID_TOUCH, + DA9034_ID_AUDIO, DA9034_ID_BUCK1, DA9034_ID_BUCK2, -- 1.6.0.4

1 0

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 03 Jun '09

03 Jun '09

The following changes since commit 203350c1a8e23adf17fd9a96d8bfc7adf63c1ff6: Mark Brown (1): ASoC: Initialise dev for the dummy S/PDIF DAI are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Cliff Cai (3): ASoC: SSM2602: remove unsupported sample rates ASoC: Blackfin: set the transfer size according the ac97_frame size ASoC: SSM2602: assign last substream to the master when shutting down Peter Ujfalusi (2): ASoC: TWL4030: Use reg_cache in twl4030_init_chip ASoC: TWL4030: Check the interface format for 4 channel mode Sonic Zhang (1): ASoC: Blackfin: document how anomaly 05000250 is handled sound/soc/blackfin/bf5xx-ac97.c | 9 +++++++++ sound/soc/blackfin/bf5xx-sport.c | 4 ++-- sound/soc/codecs/ssm2602.c | 33 +++++++++++++++++++-------------- sound/soc/codecs/twl4030.c | 16 +++++++++++----- 4 files changed, 41 insertions(+), 21 deletions(-)

2 1

[alsa-devel] [PATCH - errata corrige for quirk 1/1]
by me＠borgia.bo.it 03 Jun '09

03 Jun '09

From: Andrea Borgia <andrea(a)borgia.bo.it> Cut'n'paste mistake, whose likely result was nothing at all. Correct version is "USB_DEVICE", not "USB_DEVICE_VENDOR_SPEC". Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 58f24f5..f0f7624 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1986,7 +1986,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0028), + USB_DEVICE(0x0ccd, 0x0028), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "TerraTec", .product_name = "Aureon 5.1 MkII", -- 1.6.0.4

2 1

Re: [alsa-devel] ALSA patch Nvidia 8 channel hdmi audio
by Marek Vasut 03 Jun '09

03 Jun '09

I'm not the author of the patch, CCing alsa-devel and the original author. On Tuesday 02 of June 2009 23:58:26 Alan Alan wrote: > Hi Marek ! > > I just read your emailing with nvidia devs about alsa supporting 8 channel > lpcm over hdmi. > > I checked it out an i have to tell it is working awsome, except for a few > bugs: > > The channels are wrong mapped. Running a speakertest -Dhdmi -c6 gives the > following result (first where it should sound) > FL - FL > C - RL > FR - FR > RR - LFE > RL - C > > The other thing i realized that playback is limited to 16 bits 48 khz. I > think the minimum which is supported by nvidia hardware is 24bits 96 (or > 192) khz. > Why not ? > > Could you contact the devs as i don't know how to reach them. > > ps.: Mybe my build methot was not 100% ok ? I used 1.020 Alsa with the > oatched nvidia.c file. Anyway the amplifier shows multichannel LPCM and > that's great news.

1 0

[alsa-devel] problem with snd_hda_intel for toshiba nb200 (no audio in ubuntu remix)
by sbuonann＠libero.it 02 Jun '09

02 Jun '09

the output of lspci -nv is the fellow: 00:00.0 0600: 8086:27ac (rev 03) Subsystem: 1179:ff60 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information <?> Kernel driver in use: agpgart-intel Kernel modules: intel-agp 00:02.0 0300: 8086:27ae (rev 03) Subsystem: 1179: ff60 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at f0000000 (32- bit, non-prefetchable) [size=512K] I/O ports at 1800 [size=8] Memory at d0000000 (32-bit, prefetchable) [size=256M] Memory at f0200000 (32-bit, non- prefetchable) [size=256K] Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [d0] Power Management version 2 Kernel modules: intelfb 00:02.1 0380: 8086:27a6 (rev 03) Subsystem: 1179:ff60 Flags: bus master, fast devsel, latency 0 Memory at f0080000 (32-bit, non- prefetchable) [size=512K] Capabilities: [d0] Power Management version 2 00:1b. 0 0403: 8086:27d8 (rev 02) Subsystem: 1179:ff6e Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0440000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel 00:1c.0 0604: 8086:27d0 (rev 02) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 Capabilities: [40] Express Root Port (Slot+), MSI 00 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 1179:ff60 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> Kernel driver in use: pcieport-driver Kernel modules: shpchp 00:1c.1 0604: 8086:27d2 (rev 02) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 Memory behind bridge: f0100000-f01fffff Capabilities: [40] Express Root Port (Slot+), MSI 00 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 1179:ff60 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> Kernel driver in use: pcieport- driver Kernel modules: shpchp 00:1c.2 0604: 8086:27d4 (rev 02) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=04, subordinate=04, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: 50000000-500fffff Prefetchable memory behind bridge: 00000000f0500000- 00000000f05fffff Capabilities: [40] Express Root Port (Slot+), MSI 00 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 1179:ff60 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> Kernel driver in use: pcieport-driver Kernel modules: shpchp 00:1c.3 0604: 8086:27d6 (rev 02) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=05, subordinate=05, sec-latency=0 Capabilities: [40] Express Root Port (Slot+), MSI 00 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 1179: ff60 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> Kernel driver in use: pcieport-driver Kernel modules: shpchp 00:1d.0 0c03: 8086:27c8 (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 23 I/O ports at 1820 [size=32] Kernel driver in use: uhci_hcd 00:1d.1 0c03: 8086: 27c9 (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 1840 [size=32] Kernel driver in use: uhci_hcd 00:1d.2 0c03: 8086:27ca (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1860 [size=32] Kernel driver in use: uhci_hcd 00:1d.3 0c03: 8086:27cb (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 16 I/O ports at 1880 [size=32] Kernel driver in use: uhci_hcd 00:1d.7 0c03: 8086:27cc (rev 02) (prog-if 20) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 23 Memory at f0444000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port: BAR=1 offset=00a0 Kernel driver in use: ehci_hcd 00:1e.0 0604: 8086:2448 (rev e2) (prog-if 01) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=06, subordinate=06, sec-latency=0 Capabilities: [50] Subsystem: 1179:ff60 00:1f.0 0601: 8086:27b9 (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0 Capabilities: [e0] Vendor Specific Information <?> Kernel modules: iTCO_wdt, intel-rng 00:1f.1 0101: 8086:27df (rev 02) (prog-if 8a [Master SecP PriP]) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 1810 [size=16] Kernel driver in use: ata_piix 00:1f.2 0106: 8086:27c5 (rev 02) (prog-if 01) Subsystem: 1179:ff60 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 2299 I/O ports at 18c8 [size=8] I/O ports at 18c0 [size=4] I/O ports at 18a8 [size=8] I/O ports at 180c [size=4] I/O ports at 18b0 [size=16] Memory at f0444400 (32-bit, non-prefetchable) [size=1K] Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [70] Power Management version 2 Kernel driver in use: ahci 00:1f.3 0c05: 8086:27da (rev 02) Subsystem: 1179:ff60 Flags: medium devsel, IRQ 10 I/O ports at 18e0 [size=32] Kernel modules: i2c-i801 03:00.0 0280: 168c:002b (rev 01) Subsystem: 1113:e811 Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at f0100000 (64-bit, non-prefetchable) [size=64K] Capabilities: [40] Power Management version 3 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [60] Express Legacy Endpoint, MSI 00 Capabilities: [100] Advanced Error Reporting <?> Capabilities: [140] Virtual Channel <?> Capabilities: [160] Device Serial Number 12-14-24-ff-ff-17-15-00 Capabilities: [170] Power Budgeting <?> Kernel driver in use: ath9k Kernel modules: ath9k 04:00.0 0200: 10ec:8136 (rev 02) Subsystem: 1179:ff60 Flags: bus master, fast devsel, latency 0, IRQ 2298 I/O ports at 2000 [size=256] Memory at f0510000 (64-bit, prefetchable) [size=4K] Memory at f0500000 (64- bit, prefetchable) [size=64K] [virtual] Expansion ROM at f0520000 [disabled] [size=128K] Capabilities: [40] Power Management version 3 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [70] Express Endpoint, MSI 01 Capabilities: [ac] MSI-X: Enable- Mask- TabSize=2 Capabilities: [cc] Vital Product Data <?> Capabilities: [100] Advanced Error Reporting <?> Capabilities: [140] Virtual Channel <?> Capabilities: [160] Device Serial Number 00-00-00-00-00-00-00-01 Kernel driver in use: r8169 Kernel modules: r8169

1 0

[alsa-devel] How can i solve the alsa underrun issue
by Ren Brian 02 Jun '09

02 Jun '09

Hi, I'm trying to develop one application using alsa lib. The application receives encoded sound data from a udp socket and decodes the data and write it to device. Is it possible to set the buffer length for alsa?. In my application i have a short array of size 6000 to keep the decoded data and decoded data length is 5120. What period size i should set. It always set 940 frames. When i used the OSS calls, it works fine. When i used the alsa lib it gives underrun. How can i resolve this? Thanks, -Renju

1 1

[alsa-devel] HDAudio volume/widget control?
by pl bossart 02 Jun '09

02 Jun '09

Hi all, Bear with my ignorance here. It's not clear to me how an application or an audio server can precisely control volumes if the hierarchy of HDAudio widgets is not exposed. I checked in the HDaudio code, and it seems the connection list and connectivity entries are only showed in sysfs, not really directly usable. My findings are backed by the 2008 LPC report at http://lwn.net/Articles/299093/ which asserts that "The new audio standard, HD-Audio, is commonly found on new systems, and it's well-supported at the kernel level. However, it's based on "widgets" with vendor-configurable I/O pins. A driver can't tell how the HD-Audio part is connected, so some Linux plumbing work is required to identify which of the many exposed level controls is the right one to show the user. An audience member pointed out the need to tweak multiple level settings on his hardware, to get the right level without distortion. Linux will need more information on how each machine has its HD-Audio hardware hooked up in order to reliably give the user a useful volume control." Am I missing something? Or is there already a plan to change this? Thanks for your feedback. Cheers, Pierre

1 0

[alsa-devel] unknown card works
by Oscon 02 Jun '09

02 Jun '09

Hello! I have got a "retro" :) PCI analog tuner card. id: unknown card 0x878-0x144f:0x3000 board name: Askey Computer Corp. MagicTView'99 digital audio rate: 48000 Hz (?) ------------------ Other informations: / dmesg, modules.conf, lspci, system information (version of kernel+alsa-utils) dmesg: May 30 14:19:51 osconsfortress kernel: unknown card 0x878-0x144f:0x3000 May 30 14:19:51 osconsfortress kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> May 30 14:19:51 osconsfortress kernel: Bt87x 0000:03:06.1: PCI INT A -> GSI 21 (level, low) -> IRQ 21 May 30 14:19:51 osconsfortress kernel: bt87x0: Using board 0, analog, digital (rate 48000 Hz) -------------------- modules.conf: options snd-bt87x index=2 enable=1 load_all=1 digital_rate=48000 -------------------- lspci -vvv 03:06.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 02) Subsystem: Askey Computer Corp. MagicTView CPH060 - Audio Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (1000ns min, 63750ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at f8ffe000 (32-bit, prefetchable) [size=4K] Kernel driver in use: Bt87x --------------------------- System information: Debian Lenny kernel: root@osconsfortress:~# uname -a Linux osconsfortress 2.6.27-limbo #1 SMP Thu Mar 5 11:18:25 CET 2009 i686 GNU/Linux root@osconsfortress:~# (based on 2.6.27.10+grsec+layer7) root@osconsfortress:~# uptime 14:32:12 up 48 days, 19:41, 3 users, load average: 0.28, 0.30, 0.35 alsa-base :Version: 1.0.17.dfsg-4 -------------------- Cheers, Oscon

2 1

[alsa-devel] Playback sound is only audible starting at 50% on Dell Inspiron 1420 (Ubuntu 32-bit)
by komputes 02 Jun '09

02 Jun '09

Hello ALSA Developers, I would like to know if anyone can help me correct the following issue. Playback sound is only audible starting at 50% on Dell Inspiron 1420 running Ubuntu 9.04 Desktop x86. This is true with 9.04, 8.10, 8.04 (all x86) but does not affect Ubuntu 9.04 AMD64 on which sound is audible at ~4%. Both of these volume levels refer to the "Master" level, all other playback levels are at 100% (even the ones that are invisible/disabled by default). I hang out on #alsa where I was proposed to start alsa with the options "options snd-hda-intel model=dell-3stack" then "options snd-hda-intel model=3stack" none of which corrected this 50% volume issue. This issue as well as all technical details can be found here (includes alsa-info output): https://bugs.edge.launchpad.net/ubuntu/+source/alsa-driver/+bug/382488 Thank you for your time, -komputes

2 1

[alsa-devel] [PATCH 1/2] ALSA: snd_usb_caiaq: give better shortname
by Daniel Mack 02 Jun '09

02 Jun '09

If not passed as module option, provide an own card ID with the newly introduced snd_set_card_id() call. This will prevent ALSA from calling choose_default_name() which only takes the last part of a name containing whitespaces. This for example caused 'Audio 4 DJ' to be shortened to 'DJ', which was not very descriptive. The implementation now takes the short name and removes all whitespaces from it which is much nicer. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/device.c | 21 +++++++++++++++++++-- 1 files changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 15052e4..43bd0db 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -382,10 +382,10 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) static int __devinit init_card(struct snd_usb_caiaqdev *dev) { - char usbpath[32]; + char *c, usbpath[32]; struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; - int err; + int err, len; if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); @@ -427,6 +427,23 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); + /* if the id was not passed as module option, fill it with a shortened + * version of the product string which does not contain any + * whitespaces */ + + if (*card->id == '\0') { + char id[sizeof(card->id)]; + + memset(id, 0, sizeof(id)); + + for (c = card->shortname, len = 0; + *c && len < sizeof(card->id); c++) + if (*c != ' ') + id[len++] = *c; + + snd_card_set_id(card, id); + } + usb_make_path(usb_dev, usbpath, sizeof(usbpath)); snprintf(card->longname, sizeof(card->longname), "%s %s (%s)", -- 1.6.3.1

2 4

[alsa-devel] [PATCH 1/5] ALSA: snd_usb_caiaq: clean whitespaces
by Daniel Mack 02 Jun '09

02 Jun '09

Cosmetic changes only, no code change. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 88 +++++++++++++++++++++++----------------------- sound/usb/caiaq/device.c | 60 ++++++++++++++++---------------- sound/usb/caiaq/midi.c | 24 ++++++------ 3 files changed, 86 insertions(+), 86 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b13ce76..b144513 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -42,10 +42,10 @@ (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S24_3BE, - .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 0, /* will overwrite later */ @@ -68,7 +68,7 @@ activate_substream(struct snd_usb_caiaqdev *dev, dev->sub_capture[sub->number] = sub; } -static void +static void deactivate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { @@ -118,7 +118,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) return -EPIPE; } } - + return 0; } @@ -129,7 +129,7 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) debug("%s(%p)\n", __func__, dev); if (!dev->streaming) return; - + dev->streaming = 0; for (i = 0; i < N_URBS; i++) { @@ -154,7 +154,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (all_substreams_zero(dev->sub_playback) && all_substreams_zero(dev->sub_capture)) { - /* when the last client has stopped streaming, + /* when the last client has stopped streaming, * all sample rates are allowed again */ stream_stop(dev); dev->pcm_info.rates = dev->samplerates; @@ -194,7 +194,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; debug("%s(%p)\n", __func__, substream); - + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; @@ -205,19 +205,19 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) if (dev->streaming) return 0; - + /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ for (i=0; i < ARRAY_SIZE(rates); i++) if (runtime->rate == rates[i]) dev->pcm_info.rates = 1 << i; - + snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; if (dev->spec.data_alignment == 2) bytes_per_sample++; - + bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; @@ -232,7 +232,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) ret = stream_start(dev); if (ret) return ret; - + dev->output_running = 0; wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ); if (!dev->output_running) { @@ -273,7 +273,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) return SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + return bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else return bytes_to_frames(sub->runtime, @@ -291,7 +291,7 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer }; - + static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream **subs) { @@ -333,7 +333,7 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) @@ -354,14 +354,14 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (i = 0; i < iso->actual_length;) { if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { - for (stream = 0; - stream < dev->n_streams; + for (stream = 0; + stream < dev->n_streams; stream++, i++) { if (dev->first_packet) continue; check_byte = MAKE_CHECKBYTE(dev, stream, i); - + if ((usb_buf[i] & 0x3f) != check_byte) dev->input_panic = 1; @@ -410,21 +410,21 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } if ((dev->input_panic || dev->output_panic) && !dev->warned) { - debug("streaming error detected %s %s\n", + debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); dev->warned = 1; } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, +static void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; int stream, i; - + for (i = 0; i < iso->length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; @@ -444,7 +444,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && - i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == + i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); @@ -453,7 +453,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, static void read_completed(struct urb *urb) { - struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; @@ -478,7 +478,7 @@ static void read_completed(struct urb *urb) out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; - + if (len > 0) { spin_lock(&dev->spinlock); fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); @@ -497,14 +497,14 @@ static void read_completed(struct urb *urb) out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } - + /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; urb->iso_frame_desc[frame].length = BYTES_PER_FRAME; urb->iso_frame_desc[frame].actual_length = 0; } - + urb->number_of_packets = FRAMES_PER_URB; urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); @@ -528,7 +528,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) struct usb_device *usb_dev = dev->chip.dev; unsigned int pipe; - pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? + pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE); @@ -547,25 +547,25 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) return urbs; } - urbs[i]->transfer_buffer = + urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { log("unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } - + for (frame = 0; frame < FRAMES_PER_URB; frame++) { - struct usb_iso_packet_descriptor *iso = + struct usb_iso_packet_descriptor *iso = &urbs[i]->iso_frame_desc[frame]; - + iso->offset = BYTES_PER_FRAME * frame; iso->length = BYTES_PER_FRAME; } - + urbs[i]->dev = usb_dev; urbs[i]->pipe = pipe; - urbs[i]->transfer_buffer_length = FRAMES_PER_URB + urbs[i]->transfer_buffer_length = FRAMES_PER_URB * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; @@ -589,7 +589,7 @@ static void free_urbs(struct urb **urbs) for (i = 0; i < N_URBS; i++) { if (!urbs[i]) continue; - + usb_kill_urb(urbs[i]); kfree(urbs[i]->transfer_buffer); usb_free_urb(urbs[i]); @@ -602,11 +602,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; - dev->n_audio_in = max(dev->spec.num_analog_audio_in, - dev->spec.num_digital_audio_in) / + dev->n_audio_in = max(dev->spec.num_analog_audio_in, + dev->spec.num_digital_audio_in) / CHANNELS_PER_STREAM; dev->n_audio_out = max(dev->spec.num_analog_audio_out, - dev->spec.num_digital_audio_out) / + dev->spec.num_digital_audio_out) / CHANNELS_PER_STREAM; dev->n_streams = max(dev->n_audio_in, dev->n_audio_out); @@ -619,7 +619,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) return -EINVAL; } - ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, + ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, dev->n_audio_out, dev->n_audio_in, &dev->pcm); if (ret < 0) { @@ -632,7 +632,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); - + memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware, sizeof(snd_usb_caiaq_pcm_hardware)); @@ -651,9 +651,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) break; } - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usb_caiaq_ops); - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); snd_pcm_lib_preallocate_pages_for_all(dev->pcm, @@ -662,7 +662,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); dev->data_cb_info = - kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, + kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); if (!dev->data_cb_info) @@ -672,14 +672,14 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; } - + dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { kfree(dev->data_cb_info); free_urbs(dev->data_urbs_in); return ret; } - + dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { kfree(dev->data_cb_info); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 515de1c..9be0f2e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -79,7 +79,7 @@ static struct usb_device_id snd_usb_id_table[] = { { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = USB_VID_NATIVEINSTRUMENTS, - .idProduct = USB_PID_RIGKONTROL2 + .idProduct = USB_PID_RIGKONTROL2 }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, @@ -197,7 +197,7 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, if (buffer && len > 0) memcpy(dev->ep1_out_buf+1, buffer, len); - + dev->ep1_out_buf[0] = command; return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), dev->ep1_out_buf, len+1, &actual_len, 200); @@ -208,7 +208,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, { int ret; char tmp[5]; - + switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; case 48000: tmp[0] = SAMPLERATE_48000; break; @@ -237,12 +237,12 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (ret) return ret; - - if (!wait_event_timeout(dev->ep1_wait_queue, + + if (!wait_event_timeout(dev->ep1_wait_queue, dev->audio_parm_answer >= 0, HZ)) return -EPIPE; - - if (dev->audio_parm_answer != 1) + + if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); else dev->bpp = bpp; @@ -250,8 +250,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } -int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, - int digital, int analog, int erp) +int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, + int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, @@ -262,7 +262,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) { int ret; char val[4]; - + /* device-specific startup specials */ switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): @@ -314,7 +314,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) dev->control_state, 1); break; } - + if (dev->spec.num_analog_audio_out + dev->spec.num_analog_audio_in + dev->spec.num_digital_audio_out + @@ -323,7 +323,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) if (ret < 0) log("Unable to set up audio system (ret=%d)\n", ret); } - + if (dev->spec.num_midi_in + dev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(dev); @@ -363,7 +363,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) if (devnum >= SNDRV_CARDS) return -ENODEV; - err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, sizeof(struct snd_usb_caiaqdev), &card); if (err < 0) return err; @@ -386,7 +386,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; int err, len; - + if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); return -EIO; @@ -395,19 +395,19 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_init_urb(&dev->ep1_in_urb); usb_init_urb(&dev->midi_out_urb); - usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, + usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x1), - dev->ep1_in_buf, EP1_BUFSIZE, + dev->ep1_in_buf, EP1_BUFSIZE, usb_ep1_command_reply_dispatch, dev); - usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, + usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, usb_sndbulkpipe(usb_dev, 0x1), - dev->midi_out_buf, EP1_BUFSIZE, + dev->midi_out_buf, EP1_BUFSIZE, snd_usb_caiaq_midi_output_done, dev); - + init_waitqueue_head(&dev->ep1_wait_queue); init_waitqueue_head(&dev->prepare_wait_queue); - + if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; @@ -420,10 +420,10 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_string(usb_dev, usb_dev->descriptor.iManufacturer, dev->vendor_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iProduct, dev->product_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iSerialNumber, dev->serial, CAIAQ_USB_STR_LEN); @@ -431,7 +431,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) c = strchr(dev->serial, ' '); if (c) *c = '\0'; - + strcpy(card->driver, MODNAME); strcpy(card->shortname, dev->product_name); @@ -449,18 +449,18 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) return 0; } -static int __devinit snd_probe(struct usb_interface *intf, +static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - + ret = create_card(device, &card); - + if (ret < 0) return ret; - + usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { @@ -468,7 +468,7 @@ static int __devinit snd_probe(struct usb_interface *intf, snd_card_free(card); return ret; } - + return 0; } @@ -489,10 +489,10 @@ static void snd_disconnect(struct usb_interface *intf) snd_usb_caiaq_input_free(dev); #endif snd_usb_caiaq_audio_free(dev); - + usb_kill_urb(&dev->ep1_in_urb); usb_kill_urb(&dev->midi_out_urb); - + snd_card_free(card); usb_reset_device(interface_to_usbdev(intf)); } diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 8fa8cd8..538e8c0 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -40,7 +40,7 @@ static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *subst if (!dev) return; - + dev->midi_receive_substream = up ? substream : NULL; } @@ -64,18 +64,18 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, struct snd_rawmidi_substream *substream) { int len, ret; - + dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, EP1_BUFSIZE - 3); - + if (len <= 0) return; - + dev->midi_out_buf[2] = len; dev->midi_out_urb.transfer_buffer_length = len+3; - + ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," @@ -88,7 +88,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - + if (up) { dev->midi_out_substream = substream; if (!dev->midi_out_active) @@ -113,12 +113,12 @@ static struct snd_rawmidi_ops snd_usb_caiaq_midi_input = .trigger = snd_usb_caiaq_midi_input_trigger, }; -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len) { if (!dev->midi_receive_substream) return; - + snd_rawmidi_receive(dev->midi_receive_substream, buf, len); } @@ -142,16 +142,16 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (device->spec.num_midi_out > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usb_caiaq_midi_output); } if (device->spec.num_midi_in > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usb_caiaq_midi_input); } - + device->rmidi = rmidi; return 0; @@ -160,7 +160,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - + dev->midi_out_active = 0; if (urb->status != 0) return; -- 1.6.3.1

3 23

Re: [alsa-devel] ASoC: TWL4030 driver support multi channel playback
by Naveen Krishna Ch 02 Jun '09

02 Jun '09

2009/6/2 Mark Brown <broonie(a)opensource.wolfsonmicro.com> > On Tue, Jun 02, 2009 at 03:24:11PM +0530, Naveen Krishna Ch wrote: > > > I am done with the entire back porting of ASoC. > > > > I can see the dapm widgets getting ON/OFF and codec registers being > changed. > > > > But, there is no audio out. Am i missing somthing > > > > A direct aplay should play the audio on the headset, right? > > I am not familiar with the TWL4030, you should ask these questions on > the list. I hit reply (Instead of replyall) Sorry, > As a first step I would check that there is data being > transferred over the digital link to the CODEC, then look at the > audio routing to ensure that the signal is routed to the output. Thanks for suggestions.. -- Cheers (: Naveen Krishna Ch :)

2 1

[alsa-devel] Palm27x-asoc: fix it to work without jack gpio specified
by Marek Vasut 02 Jun '09

02 Jun '09

Hi, the patch follows. The problem was with palmz72 which doesn't have the jack detection.

2 3

[alsa-devel] ASoC: TWL4030 driver support multi channel playback
by Naveen Krishna Ch 02 Jun '09

02 Jun '09

Hi Peter, I am using linux 2.6.28 kernel, there is a requirement to support multi (4) channel play back . I guess TWL4030 ASoC driver now has full fledged support for both the DAC1(TX1) and DAC2(TX2) paths in option 1 of codec operation. I am looking forward to back port the ASoC stuff drivers from 2.6.30 rc7 to 2.6.28. Is there any better way of meeting the requirement with out back porting the stuff Suggestions please. Sorry for the english,, -- Cheers (: Naveen Krishna Ch :)

3 4

[PATCH] ALSA: introduce snd_card_make_id_unique()
by Daniel Mack 02 Jun '09

02 Jun '09

Sound card drivers may choose their own way to provide a nice sound card ID, especially when the default behaviour taken by choose_default_name() is not appropriate. ALSA's IDs must be unique, so there is need for some core logic to ensure that. This patch adds the function snd_card_make_id_unique() to the ALSA core which lowlevel drivers can call to make sure the computed string isn't already taken. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/core.h | 1 + sound/core/init.c | 53 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 54 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..01ea816 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -319,6 +319,7 @@ int snd_card_info_done(void); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); +void snd_card_make_id_unique(struct snd_card *card); #ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..bce2982 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -547,6 +547,59 @@ static void choose_default_id(struct snd_card *card) } } +static int +card_id_is_ambiguous(struct snd_card *card, const char *id) +{ + int i; + + for (i = 0; i < snd_ecards_limit; i++) { + /* only run for exisiting cards but not for the + * one in question */ + if (i == card->number || !snd_cards[i]) + continue; + + if (strcmp(snd_cards[i]->id, id) == 0) + return 1; + } + + return 0; +} + +void +snd_card_make_id_unique(struct snd_card *card) +{ + int count = 1; + char id[sizeof(card->id)]; + + if (*card->id == '\0') + return; + + /* if there's a reserved word in the card's chosen id, reset it + * to empty. choose_default_id() will then take care to generate + * an id for the card at a later point */ + if (!snd_info_check_reserved_words(card->id)) { + *card->id = '\0'; + return; + } + + strlcpy(id, card->id, sizeof(id)); + + while (card_id_is_ambiguous(card, id)) { + /* As we need to add some characters to the id, make + * sure there's enough room for it */ + int maxlen = sizeof(card->id) - (2 + (count / 10)); + + if (strlen(card->id) > maxlen) + card->id[maxlen] = '\0'; + + snprintf(id, sizeof(id), "%s_%d", card->id, count); + count++; + } + + strlcpy(card->id, id, sizeof(id)); +} +EXPORT_SYMBOL_GPL(snd_card_make_id_unique); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, -- 1.6.3.1

1 0

[PATCH] ALSA: introduce snd_card_make_id_unique()
by Daniel Mack 02 Jun '09

02 Jun '09

Sound card drivers may choose their own way to provide a nice sound card ID, especially when the default behaviour taken by choose_default_name() is not appropriate. ALSA's IDs must be unique, so there is need for some core logic to ensure that. This patch adds the function snd_card_make_id_unique() to the ALSA core which lowlevel drivers can call to make sure the computed string isn't already taken. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/core.h | 1 + sound/core/init.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..01ea816 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -319,6 +319,7 @@ int snd_card_info_done(void); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); +void snd_card_make_id_unique(struct snd_card *card); #ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..8503a01 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -547,6 +547,51 @@ static void choose_default_id(struct snd_card *card) } } +static int +card_id_is_ambiguous(struct snd_card *card, const char *id) +{ + int i; + + for (i = 0; i < snd_ecards_limit; i++) { + /* only run for exisiting cards but not for the + * one in question */ + if (i == card->number || !snd_cards[i]) + continue; + + if (strcmp(snd_cards[i]->id, id) == 0) + return 1; + } + + return 0; +} + +void +snd_card_make_id_unique(struct snd_card *card) +{ + int count = 1; + char id[sizeof(card->id)]; + + if (*card->id == '\0') + return; + + strlcpy(id, card->id, sizeof(id)); + + while (card_id_is_ambiguous(card, id)) { + /* As we need to add some characters to the id, make + * sure there's enough room for it */ + int maxlen = sizeof(card->id) - (2 + (count / 10)); + + if (strlen(card->id) > maxlen) + card->id[maxlen] = '\0'; + + snprintf(id, sizeof(id), "%s_%d", card->id, count); + count++; + } + + strlcpy(card->id, id, sizeof(id)); +} +EXPORT_SYMBOL_GPL(snd_card_make_id_unique); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, -- 1.6.3.1

1 0

[alsa-devel] [PATCH] Acer Aspire 8930G support
by Hector Martin 02 Jun '09

02 Jun '09

Getting this one to work has annoyed me to no end. Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- alsa-driver-1.0.20.8.g6254b.219.g5b2b1/alsa-kernel/pci/hda/patch_realtek_orig.c 2009-06-02 07:48:47.000000000 +0200 +++ alsa-driver-1.0.20.8.g6254b.219.g5b2b1/alsa-kernel/pci/hda/patch_realtek.c 2009-06-02 08:32:29.000000000 +0200 @@ -223,6 +223,7 @@ ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, @@ -313,6 +314,8 @@ const struct hda_channel_mode *channel_mode; int num_channel_mode; int need_dac_fix; + int const_channel_count; + int ext_channel_count; /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ @@ -368,6 +371,7 @@ unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; int need_dac_fix; + int const_channel_count; unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); @@ -462,7 +466,7 @@ struct alc_spec *spec = codec->spec; return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - spec->multiout.max_channels); + spec->ext_channel_count); } static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, @@ -472,9 +476,12 @@ struct alc_spec *spec = codec->spec; int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; + &spec->ext_channel_count); + if(err >= 0 && spec->const_channel_count == 0) { + spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + } return err; } @@ -854,8 +861,13 @@ spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; spec->need_dac_fix = preset->need_dac_fix; + spec->const_channel_count = preset->const_channel_count; - spec->multiout.max_channels = spec->channel_mode[0].channels; + if(preset->const_channel_count == 0) + spec->multiout.max_channels = preset->const_channel_count; + else + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->ext_channel_count = spec->channel_mode[0].channels; spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; @@ -1456,6 +1468,44 @@ { } }; +/* + * ALC888 Acer Aspire 8930G model + */ + +static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Connect Internal Front to Front */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Internal Rear to Rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect Internal CLFE to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect HP out to Front */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Enable all DACs */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x03}, // DAC DISABLE/MUTE 1? + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, // bits 1-5, when set, disable DAC nids 0x02-0x06 apparently. Init value 0x38 + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, // DAC DISABLE/MUTE 2? + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, // some bit here disables the other DACs. Init value 0x4900 +/* Enable amplifiers */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + { } +}; + static struct hda_input_mux alc888_2_capture_sources[2] = { /* Front mic only available on one ADC */ { @@ -1508,6 +1558,17 @@ alc_automute_amp(codec); } +static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x1b; + alc_automute_amp(codec); +} + /* * ALC880 3-stack model * @@ -8708,6 +8769,7 @@ [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", + [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", [ALC883_LAPTOP_EAPD] = "laptop-eapd", @@ -8746,6 +8808,8 @@ ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", + ALC888_ACER_ASPIRE_8930G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -8959,6 +9023,27 @@ .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_8930G] = { + .mixers = { alc888_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_8930g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .const_channel_count = 6, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_2_capture_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_8930g_init_hook, + }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, alc883_chmode_mixer },

2 5

[alsa-devel] SPDIF/AES input sample rate handling?
by Robert Hancock 02 Jun '09

02 Jun '09

A question about how ALSA drivers should handle SPDIF or AES digital audio inputs: I may be eventually be writing an ALSA driver for an audio card which has multiple AES digital audio inputs. The card would support multiple sample rates on input with the provided input data being fed directly to software with no sample rate conversion. This raises the question of how to handle requests for a particular sample rate. If the user app requests recording at a sample rate of say 32KHz and we say "OK", and then they proceed to connect an input running at 44.1KHz sampling rate, the card will produce and the app will receive samples at a faster than expected rate. Presumably this is something the app just has to deal with, but will anything in the ALSA core/library puke because of it? Or is there a better way to deal with this kind of situation?

2 1

Re: [alsa-devel] Macbook[Pro] 5 6ch support
by Takashi Iwai 02 Jun '09

02 Jun '09

At Mon, 1 Jun 2009 20:21:27 +0200, Kacper Szczesniak wrote: > > On Mon, Jun 1, 2009 at 6:54 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 1 Jun 2009 15:36:35 +0200, > > Kacper Szczesniak wrote: > >> > >> On Mon, Jun 1, 2009 at 10:54 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > Hm, this patch changes the channel mapping completely. > >> > Is it intentional? > >> yes, 0xc/0xd/0xe for Front/LFE/Surr > >> > >> >> static struct snd_kcontrol_new alc885_mb5_mixer[] = { > >> >> - HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT), > >> >> - HDA_BIND_MUTE ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT), > >> >> - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), > >> >> - HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), > >> >> HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), > >> >> HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), > >> > > >> > > >> > I'm wondering whether this breaks MB5, not MB5.1. If it's MB5.1 > >> > specific, please create another model instead of breaking the old > >> > one. > >> it's confirmed to work on MB5.1 MBP5.1 MBP52 > > > > Hrm, then I wonder why the old mb5 model works as is although the > > mapping is totally different... > > /* Front mixer */ > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > - /* LineOut mixer */ > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > - /* Front Pin: output 0 (0x0d) */ > + /* Surround mixer */ > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* LFE mixer */ > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* HP mixer */ > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* Front Pin (0x0c) */ > {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, > {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, > - /* HP Pin: output 0 (0x0c) */ > + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, > + /* LFE Pin (0x0e) */ > + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, > + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, > + /* HP Pin (0x0f) */ > {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, > {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, > + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, > > connections are different also... I've changed mixer names and connected PINs to > the right mixer nodes. Fair enough. I merged the patch now to sound git tree. Let me know if you see still any problem. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - fix audio on LG R510
by Ozan Çağlayan 02 Jun '09

02 Jun '09

Currently, LG R510 is only able to produce sound on headphones, the internal speakers are not working. The user tested and confirmed that with model=Dell headphones, internal speakers and the microphone are working flawlessly. Tested-by: Serdar Soytetir <tulliana(a)gmail.com> Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- 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 0fd258e..a2133ac 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12063,6 +12063,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), + SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), -- 1.6.2

2 2

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Additional check for 4 channel mode
by Peter Ujfalusi 02 Jun '09

02 Jun '09

Hello, the following patch adds additional check in case of four channel mode. The codec has to be configured for TDM (DSP_A) mode when using 4 channel audio. Without the patch boards which does not prepared for the four channel mode could try to play four channel audio, but configure the interface to be in I2S mode, which is not valid. --- Peter Ujfalusi (1): ASoC: TWL4030: Check the interface format for 4 channel mode sound/soc/codecs/twl4030.c | 12 +++++++++--- 1 files changed, 9 insertions(+), 3 deletions(-)

2 2

[alsa-devel] [PATCH] alsa-info.sh: Provide system manufacturer and product name from DMI
by Ozan Çağlayan 02 Jun '09

02 Jun '09

This commit adds system manufacturer and product name informations acquired using dmidecode to the output of the alsa-info script. Note that those informations will only be available when dmidecode utility is installed and alsa-info is run with root privileges. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- utils/alsa-info.sh | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index f1660f8..4be4e05 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -400,6 +400,12 @@ ESDINST=$(which esd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) PAINST=$(which pulseaudio 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) ARTSINST=$(which artsd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) JACKINST=$(which jackd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) +DMIDECODE=$(which dmidecode 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) + +if [ -x $DMIDECODE ]; then + DMI_SYSTEM_MANUFACTURER=$($DMIDECODE -s system-manufacturer 2>/dev/null) + DMI_SYSTEM_PRODUCT_NAME=$($DMIDECODE -s system-product-name 2>/dev/null) +fi cat /proc/asound/modules 2>/dev/null|awk {'print $2'}>$TEMPDIR/alsamodules.tmp cat /proc/asound/cards >$TEMPDIR/alsacards.tmp @@ -431,6 +437,13 @@ echo "" >> $FILE echo $DISTRO >> $FILE echo "" >> $FILE echo "" >> $FILE +echo "!!DMI Information" >> $FILE +echo "!!---------------" >> $FILE +echo "" >> $FILE +echo "Manufacturer: $DMI_SYSTEM_MANUFACTURER" >> $FILE +echo "Product Name: $DMI_SYSTEM_PRODUCT_NAME" >> $FILE +echo "" >> $FILE +echo "" >> $FILE echo "!!Kernel Information" >> $FILE echo "!!------------------" >> $FILE echo "" >> $FILE -- 1.6.3

2 3