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

[alsa-devel] PATCH: Fix snd-hda-intel boot hang on ASUS A8N-VM
by Brian Lindholm 21 Mar '10

21 Mar '10

This patch fixes a regression in 2.6.33.x. Was working in 2.6.32.x. Symptom: Hang on boot at "HDA Intel". Boots only if on-board sound is disabled or if "snd-hda-intel.enable_msi=0" is appended to boot parameters. Machine: ASUS A8N-VM with Athlon-X2 processor. Relevant output from lspci -nn: 00:10.1 Audio device [0403]: nVidia Corporation MCP51 High Definition Audio [10de:026c] (rev a2) Relevant output from alsa-info.sh: 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) 00:10.1 0403: 10de:026c (rev a2) Subsystem: 10de:cb84 Proposed patch (which has been confirmed to allow boot, with sound active): diff -dru linux-2.6.33-orig/sound/pci/hda/hda_intel.c linux-2.6.33/sound/pci/hda/hda_intel.c --- linux-2.6.33-orig/sound/pci/hda/hda_intel.c 2010-03-16 17:33:23.487938000 -0400 +++ linux-2.6.33/sound/pci/hda/hda_intel.c 2010-03-16 17:40:36.141225010 -0400 @@ -2354,6 +2354,7 @@ static void __devinit check_probe_mask(s static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ + SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS", 0), /* nvidia */ {} }; Signed-off-by: Brian E Lindholm <brian_lindholm(a)users.sourceforge.net>

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[alsa-devel] question about sound/oss/vidc.c
by Julia Lawall 20 Mar '10

20 Mar '10

In sound/oss/vidc.c, the function vidc_audio_trigger contains thefollowing code: struct audio_operations *adev = audio_devs[dev]; if (enable_bits & PCM_ENABLE_OUTPUT) { if (!(adev->flags & DMA_ACTIVE)) { unsigned long flags; local_irq_save(flags); /* prevent recusion */ adev->flags |= DMA_ACTIVE; ... } } I wonder if the use of DMA_ACTIVE is correct? This constant is defined inthe file sound/oss/dev_table.h within the definition of the structure dma_buffparms. But here it is being used with the structure audio_operations. The definition of audio_operations also contains the definitions of some constants. Indeed, one of them, DMA_DUPLEX has the same value as DMA_ACTIVE. thanks, julia

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[alsa-devel] Problem about building alsa driver for wm8974 on i.mx27 platform
by jjinfo＠nudt.edu.cn 20 Mar '10

20 Mar '10

Hi, I'm trying to build alsa-1.0.22 driver for my wm8974 codec on i.Mx27 platform. I need to cross compile it on x86 platform for i.Mx27. Here is my configure options: ./configure --wit-cross=arm-linux- --with-kernel=<path-to-ARM-kernel> --with-build=<path-to-kernel> When configure process completes, a "toplevel.config" file is generated. But in the toplevel.config file, the following config options are not set by the configure process( which prevent wm8974 driver from being built): CONFIG_SND_MX1_MX2_SOC= CONFIG_ARCH_MX2= CONFIG_ARCH_MX1= CONFIG_SND_MXC_SOC_SSI= CONFIG_SND_SOC_MX27VIS_WM8974= CONFIG_MACH_MX27= CONFIG_MACH_IMX27_VISSTRIM_M10= Since the default value of "--with-cards" is "all", why these options are not set? (Event I specifiy --with-cards=soc, mx1-mx2-soc, soc-mx27vis-wm8974, these options are still not set in toplevel.config.) Any idea? Regards, Jie

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[alsa-devel] Question about support for TLV320AIC3104 codec
by jjinfo＠nudt.edu.cn 20 Mar '10

20 Mar '10

Hi all, Does the codec driver of tlv320aic3x.c/.h in the latest release of alsa-1.0.22 support TLV320AIC3204 codec? Regards, Jie

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[PATCH] ASoC: enable wm8753 in aspenite
by Haojian Zhuang 19 Mar '10

19 Mar '10

Enable wm8753 codec in aspenite via ssp interface. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/aspenite.c | 190 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 201 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/aspenite.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 286d52a..0f10ee2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -113,6 +113,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_ASPENITE + tristate "SoC Audio support for Marvell Aspenite" + depends on SND_PXA2XX_SOC && MACH_ASPENITE + select SND_PXA168_SOC_SSP + select SND_SOC_WM8753 + help + Say Y if you want to add support for SoC audio on the + Marvell Aspenite reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index a74e6c9..74054aa 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -12,6 +12,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o obj-$(CONFIG_SND_PXA168_SOC_SSP) += snd-soc-pxa168-ssp.o # PXA Machine Support +snd-soc-aspenite-objs := aspenite.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o +obj-$(CONFIG_SND_SOC_ASPENITE) += snd-soc-aspenite.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/aspenite.c b/sound/soc/pxa/aspenite.c new file mode 100644 index 0000000..f82acdc --- /dev/null +++ b/sound/soc/pxa/aspenite.c @@ -0,0 +1,190 @@ +/* + * aspenite.c -- SoC audio for Aspenite + * + * Copyright (C) 2009-2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/ssp.h> + +#include "../codecs/wm8753.h" +#include "pxa2xx-pcm.h" +#include "pxa168-ssp.h" + +static struct snd_soc_card aspenite; + +/* aspenite machine dapm widgets */ +static const struct snd_soc_dapm_widget aspenite_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Headset Speaker L", NULL), + SND_SOC_DAPM_SPK("Headset Speaker R", NULL), + SND_SOC_DAPM_LINE("Line LIN", NULL), + SND_SOC_DAPM_LINE("Line RIN", NULL), +}; + +/* aspenite machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone connected to LOUT1/ROUT1 */ + {"Headphone", NULL, "LOUT1"}, + {"Headphone", NULL, "ROUT1"}, + + /* Speaker connected to LOUT2/OUT4 & OUT3/ROUT2 */ + {"Headset Speaker L", NULL, "LOUT2"}, + {"Headset Speaker L", NULL, "OUT4"}, + {"Headset Speaker R", NULL, "OUT3"}, + {"Headset Speaker R", NULL, "ROUT2"}, + + /* Line connected to LINE1/LINE2 */ + {"Line LIN", NULL, "LINE1"}, + {"Line RIN", NULL, "LINE2"}, + + /* Mic */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const struct snd_kcontrol_new wm8753_aspenite_controls[] = { + SOC_DAPM_PIN_SWITCH("Headset Speaker L"), + SOC_DAPM_PIN_SWITCH("Headset Speaker R"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Line LIN"), + SOC_DAPM_PIN_SWITCH("Line RIN"), +}; + +static int aspenite_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int mclk, ret; + + mclk = pxa168_query_mclk(substream, params); + if (mclk < 0) + return mclk; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aspenite_hifi_ops = { + .hw_params = aspenite_hifi_hw_params, +}; + +static int aspenite_wm8753_init(struct snd_soc_codec *codec) +{ + int ret; + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "MONO1"); + snd_soc_dapm_nc_pin(codec, "MONO2"); + snd_soc_dapm_nc_pin(codec, "RXP"); + snd_soc_dapm_nc_pin(codec, "RXN"); + + snd_soc_dapm_new_controls(codec, aspenite_dapm_widgets, + ARRAY_SIZE(aspenite_dapm_widgets)); + ret = snd_soc_add_controls(codec, wm8753_aspenite_controls, + ARRAY_SIZE(wm8753_aspenite_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +static struct snd_soc_dai_link aspenite_dai[] = { + { + .name = "WM8753 HiFi", + .stream_name = "WM8753 HiFi", + .cpu_dai = &pxa168_ssp_dai[PXA168_DAI_SSP1], + .codec_dai = &wm8753_dai[0], + .init = aspenite_wm8753_init, + .ops = &aspenite_hifi_ops, + }, +}; + +static struct snd_soc_card aspenite = { + .name = "Aspenite", + .platform = &pxa2xx_soc_platform, + .dai_link = aspenite_dai, + .num_links = ARRAY_SIZE(aspenite_dai), +}; + +static struct snd_soc_device aspenite_snd_devdata = { + .card = &aspenite, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static struct platform_device *aspenite_snd_device; + +static int __init aspenite_init(void) +{ + int ret; + + aspenite_snd_device = platform_device_alloc("soc-audio", -1); + if (!aspenite_snd_device) + return -ENOMEM; + + platform_set_drvdata(aspenite_snd_device, &aspenite_snd_devdata); + aspenite_snd_devdata.dev = &aspenite_snd_device->dev; + + ret = platform_device_add(aspenite_snd_device); + if (ret) + platform_device_put(aspenite_snd_device); + return ret; +} +module_init(aspenite_init); + +static void __exit aspenite_exit(void) +{ + platform_device_unregister(aspenite_snd_device); +} +module_exit(aspenite_exit); + +MODULE_DESCRIPTION("ALSA SoC WM8753 Aspenite"); +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_LICENSE("GPL"); -- 1.5.6.5

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[PATCH] ASoC: enable wm8753 in aspenite
by Haojian Zhuang 19 Mar '10

19 Mar '10

Enable wm8753 codec in aspenite via ssp interface. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/aspenite.c | 218 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 229 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/aspenite.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 286d52a..0f10ee2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -113,6 +113,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_ASPENITE + tristate "SoC Audio support for Marvell Aspenite" + depends on SND_PXA2XX_SOC && MACH_ASPENITE + select SND_PXA168_SOC_SSP + select SND_SOC_WM8753 + help + Say Y if you want to add support for SoC audio on the + Marvell Aspenite reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index a74e6c9..74054aa 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -12,6 +12,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o obj-$(CONFIG_SND_PXA168_SOC_SSP) += snd-soc-pxa168-ssp.o # PXA Machine Support +snd-soc-aspenite-objs := aspenite.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o +obj-$(CONFIG_SND_SOC_ASPENITE) += snd-soc-aspenite.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/aspenite.c b/sound/soc/pxa/aspenite.c new file mode 100644 index 0000000..733c95c --- /dev/null +++ b/sound/soc/pxa/aspenite.c @@ -0,0 +1,218 @@ +/* + * aspenite.c -- SoC audio for Aspenite + * + * Copyright (C) 2009-2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/ssp.h> + +#include "../codecs/wm8753.h" +#include "pxa2xx-pcm.h" +#include "pxa168-ssp.h" + +static struct snd_soc_card aspenite; + +/* aspenite machine dapm widgets */ +static const struct snd_soc_dapm_widget aspenite_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Headset Speaker L", NULL), + SND_SOC_DAPM_SPK("Headset Speaker R", NULL), + SND_SOC_DAPM_LINE("Line LIN", NULL), + SND_SOC_DAPM_LINE("Line RIN", NULL), +}; + +/* aspenite machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone connected to LOUT1/ROUT1 */ + {"Headphone", NULL, "LOUT1"}, + {"Headphone", NULL, "ROUT1"}, + + /* Speaker connected to LOUT2/OUT4 & OUT3/ROUT2 */ + {"Headset Speaker L", NULL, "LOUT2"}, + {"Headset Speaker L", NULL, "OUT4"}, + {"Headset Speaker R", NULL, "OUT3"}, + {"Headset Speaker R", NULL, "ROUT2"}, + + /* Line connected to LINE1/LINE2 */ + {"Line LIN", NULL, "LINE1"}, + {"Line RIN", NULL, "LINE2"}, + + /* Mic */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const struct snd_kcontrol_new wm8753_aspenite_controls[] = { + SOC_DAPM_PIN_SWITCH("Headset Speaker L"), + SOC_DAPM_PIN_SWITCH("Headset Speaker R"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Line LIN"), + SOC_DAPM_PIN_SWITCH("Line RIN"), +}; + +/* Seek the index of MCLK configuration table */ +static int pxa168_seek_mclk_conf(int rate, int format, int channel) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mclk_conf); i++) { + if ((mclk_conf[i].rate == rate) + && (mclk_conf[i].format == format) + && (mclk_conf[i].channel == channel)) + return i; + } + return -EINVAL; +} + +/* Get the MCLK frequency */ +static int pxa168_get_mclk(int i) +{ + if ((i < 0) || (i >= ARRAY_SIZE(mclk_conf))) + return -EINVAL; + return mclk_conf[i].mclk; +} + +static int aspenite_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int rate, width, channel; + int index, mclk, ret; + + rate = params_rate(params); + width = snd_pcm_format_physical_width(params_format(params)); + channel = params_channels(params); + ret = pxa168_seek_mclk_conf(rate, width, channel); + if (ret < 0) + return ret; + index = ret; + mclk = pxa168_get_mclk(ret); + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, index, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aspenite_hifi_ops = { + .hw_params = aspenite_hifi_hw_params, +}; + +static int aspenite_wm8753_init(struct snd_soc_codec *codec) +{ + int ret; + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "MONO1"); + snd_soc_dapm_nc_pin(codec, "MONO2"); + snd_soc_dapm_nc_pin(codec, "RXP"); + snd_soc_dapm_nc_pin(codec, "RXN"); + + snd_soc_dapm_new_controls(codec, aspenite_dapm_widgets, + ARRAY_SIZE(aspenite_dapm_widgets)); + ret = snd_soc_add_controls(codec, wm8753_aspenite_controls, + ARRAY_SIZE(wm8753_aspenite_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +static struct snd_soc_dai_link aspenite_dai[] = { + { + .name = "WM8753 HiFi", + .stream_name = "WM8753 HiFi", + .cpu_dai = &pxa168_ssp_dai[PXA168_DAI_SSP1], + .codec_dai = &wm8753_dai[0], + .init = aspenite_wm8753_init, + .ops = &aspenite_hifi_ops, + }, +}; + +static struct snd_soc_card aspenite = { + .name = "Aspenite", + .platform = &pxa2xx_soc_platform, + .dai_link = aspenite_dai, + .num_links = ARRAY_SIZE(aspenite_dai), +}; + +static struct snd_soc_device aspenite_snd_devdata = { + .card = &aspenite, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static struct platform_device *aspenite_snd_device; + +static int __init aspenite_init(void) +{ + int ret; + + aspenite_snd_device = platform_device_alloc("soc-audio", -1); + if (!aspenite_snd_device) + return -ENOMEM; + + platform_set_drvdata(aspenite_snd_device, &aspenite_snd_devdata); + aspenite_snd_devdata.dev = &aspenite_snd_device->dev; + + ret = platform_device_add(aspenite_snd_device); + if (ret) + platform_device_put(aspenite_snd_device); + return ret; +} +module_init(aspenite_init); + +static void __exit aspenite_exit(void) +{ + platform_device_unregister(aspenite_snd_device); +} +module_exit(aspenite_exit); + +MODULE_DESCRIPTION("ALSA SoC WM8753 Aspenite"); +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_LICENSE("GPL"); -- 1.5.6.5

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 19 Mar '10

19 Mar '10

The following changes since commit f4bee1bb0083869b1b7d73c6eda7703a7aae0506: Barry Song (1): ASoC: soc-cache: let reg be AND'ed by 0xff instead of data buffer for 8_8 mode are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Barry Song (2): ASoC: rename ad1938 to ad193x and add support for ad1936/7/8/9 ASoC: change bf5xx-ad1938 machine driver to bf5xx-ad193x machine driver Guennadi Liakhovetski (2): ASoC: SIU driver shall select FW_LOADER ASoC: remove a card from the list, if instantiation failed Jorge Eduardo Candelaria (1): ASoC: McPDM: Use tabs for indentation Mark Brown (9): ALSA: Rename jack switch table in preparation for button support ALSA: Add support for key reporting via the jack interface ASoC: Correct typoed Mic2 connections on 1133-EV1 board ASoC: Bail out of wm_hubs DC servo if calibration fails ASoC: Unexport AD193x bus probe/remove functions Merge branch 'topic/jack' into for-2.6.35 ASoC: Hook up microphone jack detection on 1133-EV1 board ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream Merge branch 'for-2.6.34' into for-2.6.35 Misael Lopez Cruz (1): ASoC: TWL6040: Add twl6040 codec driver Olaya, Margarita (1): OMAP4: PMIC: Rename twl6030_codec as twl6040_codec Peter Ujfalusi (3): ASoC: omap-mcbsp: Add support for Left Justified format ASoC: tlv320dac33: Fix DSP modes ASoC: tlv320dac33: Internal clocking changes drivers/mfd/twl-core.c | 4 +- include/linux/i2c/twl.h | 6 +- include/sound/jack.h | 8 + include/sound/soc-dai.h | 18 +- include/sound/soc.h | 1 + sound/core/jack.c | 71 +- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 6 +- sound/soc/blackfin/Kconfig | 9 +- sound/soc/blackfin/Makefile | 4 +- .../blackfin/{bf5xx-ad1938.c => bf5xx-ad193x.c} | 66 +- sound/soc/codecs/Kconfig | 8 +- sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/ad1938.c | 521 --------- sound/soc/codecs/ad1938.h | 100 -- sound/soc/codecs/ad193x.c | 558 +++++++++ sound/soc/codecs/ad193x.h | 81 ++ sound/soc/codecs/tlv320dac33.c | 10 +- sound/soc/codecs/twl6040.c | 1227 ++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++ sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/davinci/davinci-i2s.c | 3 +- sound/soc/davinci/davinci-mcasp.c | 3 +- sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/davinci/davinci-vcif.c | 3 +- sound/soc/imx/imx-pcm-dma-mx2.c | 8 +- sound/soc/imx/imx-ssi.c | 7 +- sound/soc/imx/wm1133-ev1.c | 19 +- sound/soc/omap/mcpdm.c | 548 +++++----- sound/soc/omap/omap-mcbsp.c | 16 +- sound/soc/omap/omap-mcpdm.c | 3 +- sound/soc/omap/omap-pcm.c | 4 +- sound/soc/pxa/pxa-ssp.c | 23 +- sound/soc/pxa/pxa2xx-ac97.c | 17 +- sound/soc/pxa/pxa2xx-i2s.c | 7 +- sound/soc/pxa/pxa2xx-pcm.c | 4 +- sound/soc/s3c24xx/s3c-ac97.c | 21 +- sound/soc/s3c24xx/s3c-dma.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 13 +- sound/soc/s3c24xx/s3c-pcm.c | 7 +- sound/soc/s3c24xx/s3c2412-i2s.c | 7 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 19 +- sound/soc/s6000/s6000-i2s.c | 3 +- sound/soc/s6000/s6000-pcm.c | 40 +- sound/soc/sh/Kconfig | 1 + sound/soc/soc-core.c | 27 +- 46 files changed, 2611 insertions(+), 1049 deletions(-) rename sound/soc/blackfin/{bf5xx-ad1938.c => bf5xx-ad193x.c} (65%) delete mode 100644 sound/soc/codecs/ad1938.c delete mode 100644 sound/soc/codecs/ad1938.h create mode 100644 sound/soc/codecs/ad193x.c create mode 100644 sound/soc/codecs/ad193x.h create mode 100755 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 19 Mar '10

19 Mar '10

The following changes since commit 83f40d31ca3fba3627a289e262b9423e21e1f01a: Takashi Iwai (1): Merge branch 'fix/asoc' into for-next are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Mark Brown (1): ASoC: Bail out of wm_hubs DC servo if calibration fails Peter Ujfalusi (2): ASoC: tlv320dac33: Fix DSP modes ASoC: tlv320dac33: Internal clocking changes sound/soc/codecs/tlv320dac33.c | 10 +++------- sound/soc/codecs/wm_hubs.c | 2 +- 2 files changed, 4 insertions(+), 8 deletions(-)

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[alsa-devel] [PATCH v2] ASoC: remove a card from the list, if instantiation failed
by Guennadi Liakhovetski 19 Mar '10

19 Mar '10

If instantiation of a card failed, we still have to remove it from the card list on unregistration. This fixes an Oops on Migo-R, triggering, when after a failed firmware load attempt the driver modules are removed and re-inserted again. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- Mark, is this more like what you mean? diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556..f63b59a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1232,26 +1232,25 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - if (!card->instantiated) - return 0; + if (card->instantiated) { + run_delayed_work(&card->delayed_work); - run_delayed_work(&card->delayed_work); + if (platform->remove) + platform->remove(pdev); - if (platform->remove) - platform->remove(pdev); + if (codec_dev->remove) + codec_dev->remove(pdev); - if (codec_dev->remove) - codec_dev->remove(pdev); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->remove) + cpu_dai->remove(pdev, cpu_dai); + } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->remove) - cpu_dai->remove(pdev, cpu_dai); + if (card->remove) + card->remove(pdev); } - if (card->remove) - card->remove(pdev); - snd_soc_unregister_card(card); return 0;

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[alsa-devel] Are ALSA io plugins implemented as threads or does ALSA use fork() ?
by Stefan Schoenleitner 19 Mar '10

19 Mar '10

Hello, from what I have learned so far ALSA io plugins are implemented as shared libraries with their entry point defined with the SND_PCM_PLUGIN_DEFINE_FUNC() macro. However, each plugin instance can only work for either the SND_PCM_STREAM_PLAYBACK or SND_PCM_STREAM_CAPTURE stream. Hence this is why io plugins supporting playback and capture usually set up their callback functions depending on what type of stream is used. For example in the pulseaudio io plugin, this is done with: pcm->io.callback = stream == SND_PCM_STREAM_PLAYBACK ? &pulse_playback_callback : &pulse_capture_callback; * If such a plugin is used for capture AND playback at the same time, it seems that two instances of the plugin need to be running, right ? * Are these instances running as separate processes (i.e. fork()) or are they implemented as threads ? How is this implemented internally in the ALSA code ? In my plugin I'm planning that both the playback and the capture plugin instances talk to a common data processing thread over shared buffers. The reason for this is that the data processing threads needs to have access to the capture and playback samples *at the same time*. I'm guessing that the plugin instances run with different PIDs so that I would need to use common IPC mechanisms in order to let them communicate with each other, right ? Hence my first approach would be to set up a common shared memory together with a semaphore so that only one process can access the shared memory at a time. Any better ideas ? cheers, Stefan

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