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

[alsa-devel] [PATCH] ALSA: hda - Add PCI quirk for HP dv6-1110ax.
by Kunal Gangakhedkar 21 Mar '10

21 Mar '10

Adding this PCI quirk fixes the board config detection. This also fixes jack sensing by using "hp_detect=1" via properly detected board config. Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c416bb..c4be3fa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1730,6 +1730,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, "HP dv6", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061, + "HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, -- 1.6.6.60.gc2ff1

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[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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