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

[alsa-devel] [PATCH] More Aspire 8930G fixes
by Hector Martin 04 Jun '09

04 Jun '09

Enable all three capture channels, including the missing nid 7 which is the only one capable of capturing DMIC input Enable Headphone amp for the HP jack. This causes a volume boost for headphones, but does not cause any noticeable effect for light loads like other amps, so there is no need to make it configurable. Add Input Mix capture mux setting to capture the output of the playback input mux (that is, what goes out the speakers except for PCM) Hack another coef register because the stereo DMIC for some reason produces a nonstandard sum/difference signal. I found a bit to make it just use the sum signal for both channels, which makes it behave like a standard mono microphone. The stereo is useless anyway (they're 1cm apart). Tested working: Three capture channels, mic in, line in, DMIC. Tested not working: CD. Not sure why, might be unconnected in the actual hardware or a CD drive issue. Also looked at SPDIF. It appears to work (emitter lights up inside the HP out jack) but I lack a proper miniTOSLINK cable to test it. -- 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-1.0.20.10.g955ee.233.gf65b3/sound/pci/hda/patch_realtek.c.orig 2009-06-03 23:17:55.000000000 +0200 +++ alsa-driver-1.0.20.10.g955ee.233.gf65b3/sound/pci/hda/patch_realtek.c 2009-06-03 23:45:40.000000000 +0200 @@ -1469,10 +1469,10 @@ }; /* - * ALC888 Acer Aspire 8930G model + * ALC889 Acer Aspire 8930G model */ -static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +static struct hda_verb alc889_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 */ @@ -1492,7 +1492,7 @@ {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_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Enable all DACs */ @@ -1507,6 +1507,17 @@ /* Enable amplifiers */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, +/* DMIC fix + * This laptop has a stereo digital microphone. The mics are only 1cm apart + * which makes the stereo useless. However, either the mic or the ALC889 + * makes the signal become a difference/sum signal instead of standard + * stereo, which is annoying. So instead we flip this bit which makes the + * codec replicate the sum signal to both channels, turning it into a + * normal mono mic. + */ +/* DMIC_CONTROL? Init value = 0x0001 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x0b}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0003}, { } }; @@ -1531,6 +1542,38 @@ } }; +static struct hda_input_mux alc889_capture_sources[3] = { + /* Digital mic only available on first "ADC" */ + { + .num_items = 5, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Front Mic", 0xb }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + } +}; + static struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1562,7 +1605,7 @@ alc_automute_amp(codec); } -static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9081,22 +9124,22 @@ .mixers = { alc888_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc888_acer_aspire_8930g_verbs }, + alc889_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, + .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), + .adc_nids = alc889_adc_nids, + .capsrc_nids = alc889_capsrc_nids, .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, + ARRAY_SIZE(alc889_capture_sources), + .input_mux = alc889_capture_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_acer_aspire_8930g_init_hook, + .init_hook = alc889_acer_aspire_8930g_init_hook, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer,

2 1

[alsa-devel] [PATCH] burgundy: timeout message is off by one.
by Roel Kluin 04 Jun '09

04 Jun '09

Timeout message is off by one. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 45a7629..d2c0a4e 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -46,12 +46,12 @@ snd_pmac_burgundy_extend_wait(struct snd_pmac *chip) timeout = 50; while (!(in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_extend_wait: timeout #1\n"); timeout = 50; while ((in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_extend_wait: timeout #2\n"); }

2 1

[alsa-devel] [PATCH - quirk for USB Aureon cards 1/1] Add quirk to provide proper naming of the Terratec Aureon 5.1 MkII USB card
by Andrea Borgia 03 Jun '09

03 Jun '09

Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 5d955aa..d84d6f3 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1951,6 +1951,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0028), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "TerraTec", + .product_name = "Aureon 5.1 MkII", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ USB_DEVICE(0x0ccd, 0x0035), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Miditech", -- 1.6.0.4

2 8

[alsa-devel] Sequencer timestamps for soft synth app
by Glyn Adgie 03 Jun '09

03 Jun '09

I have written a soft synth, which uses ALSA to get MIDI events, and Jack to do audio output. MIDI events are fetched and processed at the start of the Jack audio callback, then the Jack output buffers are filled by the synth engine. The problem I have is that, if the Jack buffer size is too large, the time resolution for MIDI events is too coarse, and this has a severe effect on rhythms. I currently get around this by setting up Jack for a buffer size of 64 frames, and a sample rate of 48 kHz, giving a timing resolution of 1.33 msec, which appears to be adequate. However, I might not always be able to get the desired buffer size. If the Jack buffer is some multiple of 64 frames, the synth renders multiple 64 frame chunks until the buffer is filled. This takes care of audio quality. However, MIDI timing resolution suffers. I am looking at using the event's timestamp, to decide when to process an event. For example, I might have a Jack buffer of 256 frames, so I want to work out whether to process an event right away, or after one 64 frame chunk, and so on. It would appear that, in order to make this work, I need an initial timestamp, which I get at the start of the audio callback. I then compare each event's timestamp, to determine where it should be processed. I cannot find an ALSA function for getting a timestamp. Perhaps I am missing something, or trying to do the job in the wrong way. Regards Glyn

2 1

[alsa-devel] Xonar DX spdif advice request
by Dmitri Seletski 03 Jun '09

03 Jun '09

Hello guys. I have perfectly working asus xonar card. it works fine with ordinary outputs, but I would like to know, did anyone try those cards with spdif? Is it worth buying something spdif compatible, if you intend to use this card? Thank you in advance. Dmitri

1 0

[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