* ASoC updates for 2.6.30
@ 2009-01-16 17:13 Mark Brown
2009-01-16 17:15 ` [PATCH 1/4] ASoC: DaVinci: Fix SFFSDR compilation error Mark Brown
2009-01-16 17:26 ` ASoC updates for 2.6.30 Takashi Iwai
0 siblings, 2 replies; 6+ messages in thread
From: Mark Brown @ 2009-01-16 17:13 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel
The following changes since commit c0106d72b8d71696dbe9dc80e2c77d4ac63f7531:
Takashi Iwai (1):
Merge branch 'topic/asoc' into next/asoc
are available in the git repository at:
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30
Ben Nizette (1):
ASoC: atmel_pcm: Remove non-existant header
Hugo Villeneuve (1):
ASoC: DaVinci: Fix SFFSDR compilation error.
Ian Molton (3):
ASoC: Driver for the WM9705 AC97 codec.
ASoC: machine driver for Toshiba e750
ASoC: machine driver for Toshiba e800
Mark Brown (1):
Merge branch 'for-2.6.29' into for-2.6.30
arch/arm/mach-pxa/e750.c | 5 +
arch/arm/mach-pxa/include/mach/eseries-gpio.h | 10 +
sound/soc/atmel/atmel-pcm.c | 2 -
sound/soc/codecs/Kconfig | 4 +
sound/soc/codecs/Makefile | 3 +
sound/soc/codecs/wm9705.c | 410 +++++++++++++++++++++++++
sound/soc/codecs/wm9705.h | 14 +
sound/soc/davinci/davinci-sffsdr.c | 20 +-
sound/soc/pxa/Kconfig | 9 +
sound/soc/pxa/Makefile | 2 +
sound/soc/pxa/e750_wm9705.c | 189 ++++++++++++
sound/soc/pxa/e800_wm9712.c | 116 ++++++-
12 files changed, 765 insertions(+), 19 deletions(-)
create mode 100644 sound/soc/codecs/wm9705.c
create mode 100644 sound/soc/codecs/wm9705.h
create mode 100644 sound/soc/pxa/e750_wm9705.c
^ permalink raw reply [flat|nested] 6+ messages in thread* [PATCH 1/4] ASoC: DaVinci: Fix SFFSDR compilation error. 2009-01-16 17:13 ASoC updates for 2.6.30 Mark Brown @ 2009-01-16 17:15 ` Mark Brown 2009-01-16 17:15 ` [PATCH 2/4] ASoC: Driver for the WM9705 AC97 codec Mark Brown 2009-01-16 17:26 ` ASoC updates for 2.6.30 Takashi Iwai 1 sibling, 1 reply; 6+ messages in thread From: Mark Brown @ 2009-01-16 17:15 UTC (permalink / raw) To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Hugo Villeneuve From: Hugo Villeneuve <hugo@hugovil.com> Remove dependency on sffsdr_fpga_set_codec_fs() when the SFFSDR FPGA module is not selected. Signed-off-by: Hugo Villeneuve <hugo@hugovil.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> --- sound/soc/davinci/davinci-sffsdr.c | 20 +++++++++++++++++--- 1 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 4935d1b..50baef1 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -25,7 +25,9 @@ #include <asm/dma.h> #include <asm/mach-types.h> +#ifdef CONFIG_SFFSDR_FPGA #include <asm/plat-sffsdr/sffsdr-fpga.h> +#endif #include <mach/mcbsp.h> #include <mach/edma.h> @@ -43,6 +45,17 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, int fs; int ret = 0; + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + +#ifndef CONFIG_SFFSDR_FPGA + /* Without the FPGA module, the Fs is fixed at 44100 Hz */ + if (fs != 44100) { + pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); + return -EINVAL; + } +#endif + /* Set cpu DAI configuration: * CLKX and CLKR are the inputs for the Sample Rate Generator. * FSX and FSR are outputs, driven by the sample Rate Generator. */ @@ -53,12 +66,13 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Fsref can be 32000, 44100 or 48000. */ - fs = params_rate(params); - pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); +#ifndef CONFIG_SFFSDR_FPGA + return 0; +#else return sffsdr_fpga_set_codec_fs(fs); +#endif } static struct snd_soc_ops sffsdr_ops = { -- 1.5.6.5 ^ permalink raw reply related [flat|nested] 6+ messages in thread
* [PATCH 2/4] ASoC: Driver for the WM9705 AC97 codec. 2009-01-16 17:15 ` [PATCH 1/4] ASoC: DaVinci: Fix SFFSDR compilation error Mark Brown @ 2009-01-16 17:15 ` Mark Brown 2009-01-16 17:15 ` [PATCH 3/4] ASoC: machine driver for Toshiba e750 Mark Brown 0 siblings, 1 reply; 6+ messages in thread From: Mark Brown @ 2009-01-16 17:15 UTC (permalink / raw) To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Ian Molton From: Ian Molton <ian@mnementh.co.uk> This driver adds support for the wm9705 ac97 codec. The driver supports audio input and output. Signed-off-by: Ian Molton <ian@mnementh.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/wm9705.c | 410 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9705.h | 14 ++ 4 files changed, 431 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm9705.c create mode 100644 sound/soc/codecs/wm9705.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d0e0d69..cb5fcd6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8990 if I2C + 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 help @@ -144,6 +145,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8990 tristate +config SND_SOC_WM9705 + tristate + config SND_SOC_WM9712 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c4ddc9a..3664cdc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o @@ -51,5 +52,7 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +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 diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c new file mode 100644 index 0000000..cb26b6a --- /dev/null +++ b/sound/soc/codecs/wm9705.c @@ -0,0 +1,410 @@ +/* + * wm9705.c -- ALSA Soc WM9705 codec support + * + * Copyright 2008 Ian Molton <spyro@f2s.com> + * + * 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; Version 2 of the License only. + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +/* + * WM9705 register cache + */ +static const u16 wm9705_reg[] = { + 0x6150, 0x8000, 0x8000, 0x8000, /* 0x0 */ + 0x0000, 0x8000, 0x8008, 0x8008, /* 0x8 */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */ + 0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */ + 0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */ + 0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */ + 0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */ +}; + +static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = { + SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), + SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), + SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), + SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), + SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), + SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1), + SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1), + SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1), + SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1), + SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), + SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0), + SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0), + SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), +}; + +static const char *wm9705_mic[] = {"Mic 1", "Mic 2"}; +static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; + +static const struct soc_enum wm9705_enum_mic = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic); +static const struct soc_enum wm9705_enum_rec_l = + SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel); +static const struct soc_enum wm9705_enum_rec_r = + SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel); + +/* Headphone Mixer */ +static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1), + SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), + SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1), + SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1), +}; + +/* Mic source */ +static const struct snd_kcontrol_new wm9705_mic_src_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_mic); + +/* Capture source */ +static const struct snd_kcontrol_new wm9705_capture_selectl_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_rec_l); +static const struct snd_kcontrol_new wm9705_capture_selectr_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_rec_r); + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0, + &wm9705_mic_src_controls), + SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, + &wm9705_capture_selectl_controls), + SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, + &wm9705_capture_selectr_controls), + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0, + &wm9705_hp_mixer_controls[0], + ARRAY_SIZE(wm9705_hp_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_INPUT("PHONE"), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + SND_SOC_DAPM_INPUT("CDINL"), + SND_SOC_DAPM_INPUT("CDINR"), + SND_SOC_DAPM_INPUT("PCBEEP"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), +}; + +/* Audio map + * WM9705 has no switches to disable the route from the inputs to the HP mixer + * so in order to prevent active inputs from forcing the audio outputs to be + * constantly enabled, we use the mutes on those inputs to simulate such + * controls. + */ +static const struct snd_soc_dapm_route audio_map[] = { + /* HP mixer */ + {"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"}, + {"HP Mixer", "CD Playback Switch", "CD PGA"}, + {"HP Mixer", "Mic Playback Switch", "Mic PGA"}, + {"HP Mixer", "Phone Playback Switch", "Phone PGA"}, + {"HP Mixer", "Line Playback Switch", "Line PGA"}, + {"HP Mixer", NULL, "Left DAC"}, + {"HP Mixer", NULL, "Right DAC"}, + + /* mono mixer */ + {"Mono Mixer", NULL, "HP Mixer"}, + + /* outputs */ + {"Headphone PGA", NULL, "HP Mixer"}, + {"HPOUTL", NULL, "Headphone PGA"}, + {"HPOUTR", NULL, "Headphone PGA"}, + {"Line out PGA", NULL, "HP Mixer"}, + {"LOUT", NULL, "Line out PGA"}, + {"ROUT", NULL, "Line out PGA"}, + {"Mono PGA", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono PGA"}, + + /* inputs */ + {"CD PGA", NULL, "CDINL"}, + {"CD PGA", NULL, "CDINR"}, + {"Line PGA", NULL, "LINEINL"}, + {"Line PGA", NULL, "LINEINR"}, + {"Phone PGA", NULL, "PHONE"}, + {"Mic Source", "Mic 1", "MIC1"}, + {"Mic Source", "Mic 2", "MIC2"}, + {"Mic PGA", NULL, "Mic Source"}, + {"PCBEEP PGA", NULL, "PCBEEP"}, + + /* Left capture selector */ + {"Left Capture Source", "Mic", "Mic Source"}, + {"Left Capture Source", "CD", "CDINL"}, + {"Left Capture Source", "Line", "LINEINL"}, + {"Left Capture Source", "Stereo Mix", "HP Mixer"}, + {"Left Capture Source", "Mono Mix", "HP Mixer"}, + {"Left Capture Source", "Phone", "PHONE"}, + + /* Right capture source */ + {"Right Capture Source", "Mic", "Mic Source"}, + {"Right Capture Source", "CD", "CDINR"}, + {"Right Capture Source", "Line", "LINEINR"}, + {"Right Capture Source", "Stereo Mix", "HP Mixer"}, + {"Right Capture Source", "Mono Mix", "HP Mixer"}, + {"Right Capture Source", "Phone", "PHONE"}, + + {"ADC PGA", NULL, "Left Capture Source"}, + {"ADC PGA", NULL, "Right Capture Source"}, + + /* ADC's */ + {"Left ADC", NULL, "ADC PGA"}, + {"Right ADC", NULL, "ADC PGA"}, +}; + +static int wm9705_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets, + ARRAY_SIZE(wm9705_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +/* We use a register cache to enhance read performance. */ +static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + switch (reg) { + case AC97_RESET: + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return soc_ac97_ops.read(codec->ac97, reg); + default: + reg = reg >> 1; + + if (reg >= (ARRAY_SIZE(wm9705_reg))) + return -EIO; + + return cache[reg]; + } +} + +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + soc_ac97_ops.write(codec->ac97, reg, val); + reg = reg >> 1; + if (reg < (ARRAY_SIZE(wm9705_reg))) + cache[reg] = val; + + return 0; +} + +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg; + u16 vra; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return ac97_write(codec, reg, runtime->rate); +} + +#define WM9705_AC97_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) + +struct snd_soc_dai wm9705_dai[] = { + { + .name = "AC97 HiFi", + .ac97_control = 1, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .prepare = ac97_prepare, + }, + }, + { + .name = "AC97 Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + } +}; +EXPORT_SYMBOL_GPL(wm9705_dai); + +static int wm9705_reset(struct snd_soc_codec *codec) +{ + if (soc_ac97_ops.reset) { + soc_ac97_ops.reset(codec->ac97); + if (ac97_read(codec, 0) == wm9705_reg[0]) + return 0; /* Success */ + } + + return -EIO; +} + +static int wm9705_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "WM9705 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return -ENOMEM; + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(wm9705_reg); + codec->reg_cache_step = 2; + + codec->name = "WM9705"; + codec->owner = THIS_MODULE; + codec->dai = wm9705_dai; + codec->num_dai = ARRAY_SIZE(wm9705_dai); + codec->write = ac97_write; + codec->read = ac97_read; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) { + printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); + goto codec_err; + } + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + ret = wm9705_reset(codec); + if (ret) + goto reset_err; + + snd_soc_add_controls(codec, wm9705_snd_ac97_controls, + ARRAY_SIZE(wm9705_snd_ac97_controls)); + wm9705_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm9705: failed to register card\n"); + goto pcm_err; + } + + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->reg_cache); +cache_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int wm9705_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm9705 = { + .probe = wm9705_soc_probe, + .remove = wm9705_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); + +MODULE_DESCRIPTION("ASoC WM9705 driver"); +MODULE_AUTHOR("Ian Molton"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h new file mode 100644 index 0000000..d380f11 --- /dev/null +++ b/sound/soc/codecs/wm9705.h @@ -0,0 +1,14 @@ +/* + * wm9705.h -- WM9705 Soc Audio driver + */ + +#ifndef _WM9705_H +#define _WM9705_H + +#define WM9705_DAI_AC97_HIFI 0 +#define WM9705_DAI_AC97_AUX 1 + +extern struct snd_soc_dai wm9705_dai[2]; +extern struct snd_soc_codec_device soc_codec_dev_wm9705; + +#endif -- 1.5.6.5 ^ permalink raw reply related [flat|nested] 6+ messages in thread
* [PATCH 3/4] ASoC: machine driver for Toshiba e750 2009-01-16 17:15 ` [PATCH 2/4] ASoC: Driver for the WM9705 AC97 codec Mark Brown @ 2009-01-16 17:15 ` Mark Brown 2009-01-16 17:15 ` [PATCH 4/4] ASoC: machine driver for Toshiba e800 Mark Brown 0 siblings, 1 reply; 6+ messages in thread From: Mark Brown @ 2009-01-16 17:15 UTC (permalink / raw) To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Ian Molton From: Ian Molton <ian@mnementh.co.uk> This patch adds support for the wm9705 ac97 codec as used in the Toshiba e750 PDA. It includes support for powering up / down the external headphone and speaker amplifiers on this machine. Signed-off-by: Ian Molton <ian@mnementh.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> --- arch/arm/mach-pxa/e750.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e750_wm9705.c | 189 +++++++++++++++++++++++++ 5 files changed, 210 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/e750_wm9705.c diff --git a/arch/arm/mach-pxa/e750.c b/arch/arm/mach-pxa/e750.c index be1ab8e..665066f 100644 --- a/arch/arm/mach-pxa/e750.c +++ b/arch/arm/mach-pxa/e750.c @@ -133,6 +133,11 @@ static unsigned long e750_pin_config[] __initdata = { /* IrDA */ GPIO38_GPIO | MFP_LPM_DRIVE_HIGH, + /* Audio power control */ + GPIO4_GPIO, /* Headphone amp power */ + GPIO7_GPIO, /* Speaker amp power */ + GPIO37_GPIO, /* Headphone detect */ + /* PC Card */ GPIO8_GPIO, /* CD0 */ GPIO44_GPIO, /* CD1 */ diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h index efbd2aa..02b28e0 100644 --- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h +++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h @@ -45,6 +45,11 @@ /* e7xx IrDA power control */ #define GPIO_E7XX_IR_OFF 38 +/* e750 audio control GPIOs */ +#define GPIO_E750_HP_AMP_OFF 4 +#define GPIO_E750_SPK_AMP_OFF 7 +#define GPIO_E750_HP_DETECT 37 + /* ASIC related GPIOs */ #define GPIO_ESERIES_TMIO_IRQ 5 #define GPIO_ESERIES_TMIO_PCLR 19 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f82e106..b9b1a3f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -61,6 +61,15 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). +config SND_PXA2XX_SOC_E750 + tristate "SoC AC97 Audio support for e750" + depends on SND_PXA2XX_SOC && MACH_E750 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e750 PDA + config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 08a9f27..c7d4cce 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o +snd-soc-e750-objs := e750_wm9705.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o @@ -22,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o +obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c new file mode 100644 index 0000000..20fbdcf --- /dev/null +++ b/sound/soc/pxa/e750_wm9705.c @@ -0,0 +1,189 @@ +/* + * e750-wm9705.c -- SoC audio for e750 + * + * Copyright 2007 (c) Ian Molton <spyro@f2s.com> + * + * 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; version 2 ONLY. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1); + + return 0; +} + +static int e750_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E750_HP_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E750_HP_AMP_OFF, 1); + + return 0; +} + +static const struct snd_soc_dapm_widget e750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Amp", NULL, "HPOUTL"}, + {"Headphone Amp", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "Headphone Amp"}, + + {"Speaker Amp", NULL, "MONOOUT"}, + {"Speaker", NULL, "Speaker Amp"}, + + {"MIC1", NULL, "Mic (Internal)"}, +}; + +static int e750_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "LOUT"); + snd_soc_dapm_nc_pin(codec, "ROUT"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e750_dapm_widgets, + ARRAY_SIZE(e750_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e750_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e750_ac97_init, + /* use ops to check startup state */ + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e750 = { + .name = "Toshiba e750", + .platform = &pxa2xx_soc_platform, + .dai_link = e750_dai, + .num_links = ARRAY_SIZE(e750_dai), +}; + +static struct snd_soc_device e750_snd_devdata = { + .card = &e750, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e750_snd_device; + +static int __init e750_init(void) +{ + int ret; + + if (!machine_is_e750()) + return -ENODEV; + + ret = gpio_request(GPIO_E750_HP_AMP_OFF, "Headphone amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp"); + if (ret) + goto free_hp_amp_gpio; + + ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + e750_snd_device = platform_device_alloc("soc-audio", -1); + if (!e750_snd_device) { + ret = -ENOMEM; + goto free_spk_amp_gpio; + } + + platform_set_drvdata(e750_snd_device, &e750_snd_devdata); + e750_snd_devdata.dev = &e750_snd_device->dev; + ret = platform_device_add(e750_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e750_snd_device); +free_spk_amp_gpio: + gpio_free(GPIO_E750_SPK_AMP_OFF); +free_hp_amp_gpio: + gpio_free(GPIO_E750_HP_AMP_OFF); + + return ret; +} + +static void __exit e750_exit(void) +{ + platform_device_unregister(e750_snd_device); + gpio_free(GPIO_E750_SPK_AMP_OFF); + gpio_free(GPIO_E750_HP_AMP_OFF); +} + +module_init(e750_init); +module_exit(e750_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); +MODULE_DESCRIPTION("ALSA SoC driver for e750"); +MODULE_LICENSE("GPL v2"); -- 1.5.6.5 ^ permalink raw reply related [flat|nested] 6+ messages in thread
* [PATCH 4/4] ASoC: machine driver for Toshiba e800 2009-01-16 17:15 ` [PATCH 3/4] ASoC: machine driver for Toshiba e750 Mark Brown @ 2009-01-16 17:15 ` Mark Brown 0 siblings, 0 replies; 6+ messages in thread From: Mark Brown @ 2009-01-16 17:15 UTC (permalink / raw) To: Takashi Iwai; +Cc: alsa-devel, Mark Brown, Ian Molton From: Ian Molton <ian@mnementh.co.uk> This patch adds support for the wm9712 ac97 codec as used in the Toshiba e800 PDA. It includes support for powering up / down the external headphone and speaker amplifiers on this machine. Signed-off-by: Ian Molton <ian@mnementh.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> --- arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/pxa/e800_wm9712.c | 116 ++++++++++++++++++++++--- 2 files changed, 107 insertions(+), 14 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h index 02b28e0..6d6e4d8 100644 --- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h +++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h @@ -50,6 +50,11 @@ #define GPIO_E750_SPK_AMP_OFF 7 #define GPIO_E750_HP_DETECT 37 +/* e800 audio control GPIOs */ +#define GPIO_E800_HP_DETECT 81 +#define GPIO_E800_HP_AMP_OFF 82 +#define GPIO_E800_SPK_AMP_ON 83 + /* ASIC related GPIOs */ #define GPIO_ESERIES_TMIO_IRQ 5 #define GPIO_ESERIES_TMIO_PCLR 19 diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 2e3386d..78a1770 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -1,8 +1,6 @@ /* * e800-wm9712.c -- SoC audio for e800 * - * Based on tosa.c - * * Copyright 2007 (c) Ian Molton <spyro@f2s.com> * * This program is free software; you can redistribute it and/or modify it @@ -13,31 +11,96 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/device.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <asm/mach-types.h> #include <mach/pxa-regs.h> #include <mach/hardware.h> #include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static struct snd_soc_card e800; +static int e800_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E800_SPK_AMP_ON, 1); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E800_SPK_AMP_ON, 0); -static struct snd_soc_dai_link e800_dai[] = { + return 0; +} + +static int e800_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], - .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], -}, + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E800_HP_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E800_HP_AMP_OFF, 1); + + return 0; +} + +static const struct snd_soc_dapm_widget e800_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic (Internal1)", NULL), + SND_SOC_DAPM_MIC("Mic (Internal2)", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "HPOUTL"}, + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "Headphone Amp"}, + + {"Speaker Amp", NULL, "MONOOUT"}, + {"Speaker", NULL, "Speaker Amp"}, + + {"MIC1", NULL, "Mic (Internal1)"}, + {"MIC2", NULL, "Mic (Internal2)"}, +}; + +static int e800_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, e800_dapm_widgets, + ARRAY_SIZE(e800_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e800_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .init = e800_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + }, }; static struct snd_soc_card e800 = { @@ -61,6 +124,22 @@ static int __init e800_init(void) if (!machine_is_e800()) return -ENODEV; + ret = gpio_request(GPIO_E800_HP_AMP_OFF, "Headphone amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp"); + if (ret) + goto free_hp_amp_gpio; + + ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1); + if (ret) + goto free_spk_amp_gpio; + e800_snd_device = platform_device_alloc("soc-audio", -1); if (!e800_snd_device) return -ENOMEM; @@ -69,8 +148,15 @@ static int __init e800_init(void) e800_snd_devdata.dev = &e800_snd_device->dev; ret = platform_device_add(e800_snd_device); - if (ret) - platform_device_put(e800_snd_device); + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e800_snd_device); +free_spk_amp_gpio: + gpio_free(GPIO_E800_SPK_AMP_ON); +free_hp_amp_gpio: + gpio_free(GPIO_E800_HP_AMP_OFF); return ret; } @@ -78,6 +164,8 @@ static int __init e800_init(void) static void __exit e800_exit(void) { platform_device_unregister(e800_snd_device); + gpio_free(GPIO_E800_SPK_AMP_ON); + gpio_free(GPIO_E800_HP_AMP_OFF); } module_init(e800_init); @@ -86,4 +174,4 @@ module_exit(e800_exit); /* Module information */ MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); MODULE_DESCRIPTION("ALSA SoC driver for e800"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); -- 1.5.6.5 ^ permalink raw reply related [flat|nested] 6+ messages in thread
* Re: ASoC updates for 2.6.30 2009-01-16 17:13 ASoC updates for 2.6.30 Mark Brown 2009-01-16 17:15 ` [PATCH 1/4] ASoC: DaVinci: Fix SFFSDR compilation error Mark Brown @ 2009-01-16 17:26 ` Takashi Iwai 1 sibling, 0 replies; 6+ messages in thread From: Takashi Iwai @ 2009-01-16 17:26 UTC (permalink / raw) To: Mark Brown; +Cc: alsa-devel At Fri, 16 Jan 2009 17:13:47 +0000, Mark Brown wrote: > > The following changes since commit c0106d72b8d71696dbe9dc80e2c77d4ac63f7531: > Takashi Iwai (1): > Merge branch 'topic/asoc' into next/asoc > > are available in the git repository at: > > git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Thanks, pulled in. Takashi ^ permalink raw reply [flat|nested] 6+ messages in thread
end of thread, other threads:[~2009-01-16 17:26 UTC | newest] Thread overview: 6+ messages (download: mbox.gz follow: Atom feed -- links below jump to the message on this page -- 2009-01-16 17:13 ASoC updates for 2.6.30 Mark Brown 2009-01-16 17:15 ` [PATCH 1/4] ASoC: DaVinci: Fix SFFSDR compilation error Mark Brown 2009-01-16 17:15 ` [PATCH 2/4] ASoC: Driver for the WM9705 AC97 codec Mark Brown 2009-01-16 17:15 ` [PATCH 3/4] ASoC: machine driver for Toshiba e750 Mark Brown 2009-01-16 17:15 ` [PATCH 4/4] ASoC: machine driver for Toshiba e800 Mark Brown 2009-01-16 17:26 ` ASoC updates for 2.6.30 Takashi Iwai
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