December 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Staticise sst_pcm_new and sst_soc_platform_drv
by Axel Lin 14 Dec '11

14 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/mid-x86/sst_platform.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 24f9471..5b936d5 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -443,7 +443,7 @@ static void sst_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) +static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; @@ -463,7 +463,7 @@ int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) } return retval; } -struct snd_soc_platform_driver sst_soc_platform_drv = { +static struct snd_soc_platform_driver sst_soc_platform_drv = { .ops = &sst_platform_ops, .pcm_new = sst_pcm_new, .pcm_free = sst_pcm_free, -- 1.7.5.4

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[PATCH] ALSA: HDA: Use LPIB Position fix for Intel SCH Poulsbo
by David Henningsson 14 Dec '11

14 Dec '11

Several people with this chipset have reported inconsistent/sloppy values for position reporting when the DMA position buffer is used, and that setting position_fix=1 have fixed their problems. BugLink: https://bugs.launchpad.net/bugs/825709 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0746ab4..f19d7fa 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2971,7 +2971,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ + /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_BUFSIZE }, /* ICH6 */ -- 1.7.5.4 --------------050108040603000000050605--

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[alsa-devel] [PATCH] ASoC: Complete initialisation before registering Samsung PCM DAI
by Mark Brown 14 Dec '11

14 Dec '11

Otherwise there's a race where the DAI might get used without everything having been set up. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/pcm.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index beef63f..3a29c26 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -570,12 +570,6 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) } clk_enable(pcm->pclk); - ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); - if (ret != 0) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - goto err5; - } - s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + S3C_PCM_RXFIFO; s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start @@ -587,6 +581,12 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get register DAI: %d\n", ret); + goto err5; + } + return 0; err5: -- 1.7.7.3

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Re: [alsa-devel] _mmap_playback_avail() short (was: Default device XRUNs with perriods=2)
by Raymond Yau 14 Dec '11

14 Dec '11

>> Do you require the sound card give you accurate playback position ? >No. Just accurate playback (or capture). But reliably, on any system, >without hardware-specific tweaks, 'out-of-the-box'. Ideally with the >potential for reasonable latency if the user requests that. And not >necessarily only on the 'latest' Alsa versions. at 48000Hz, the hwptr of snd-ymfpci is increase to next boundary (256 frames). so you cannot get accurate playback position. this situation can be emulated with snd-hda-intel by masking the value return by pointer callback with 0xffffff00 >> snd-ymfpci only update hwptr in 5.3ms intervals >> >> it seem that snd-usb-audio update hwptr in 1ms >> >> > It seems the options are:- >> > >> > 1. Use 3 periods and accept worse latency, always filling with >> > avail_frames >> >> snd-ymfpci has periods_min 3 >> and there are sound cards with periods_max = 2 >Yes, this is the sort of thing that makes it more difficult to create a good >workaround. seem only two drivers have this limitation , so you may ask the author of those drivers whether it is feasible to change periods_max to a higher value drivers/dummy.c: .periods_max = 2, soc/samsung/idma.c: .periods_max = 2,

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[alsa-devel] [PATCH] ASoC: Staticise asoc_idma_platform
by Mark Brown 14 Dec '11

14 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/idma.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index baf97eb..2bcf758 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -410,7 +410,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) idma.lp_tx_addr = addr; } -struct snd_soc_platform_driver asoc_idma_platform = { +static struct snd_soc_platform_driver asoc_idma_platform = { .ops = &idma_ops, .pcm_new = idma_new, .pcm_free = idma_free, -- 1.7.7.3

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[alsa-devel] [PATCH v2] ASoC: Tegra machine ASoC driver for boards using ALC5332 codec
by Leon Romanovsky 13 Dec '11

13 Dec '11

At this stage only Toshiba AC100/Dynabook supported. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> -- Changes since v1: * Use module_platform_driver. * Move controls, dapm_widgets and dapm_routes to snd_soc_card structure. * Set dai_fmt in snd_soc_dai_link structure instead probe function. * Remove check board in probe function * Fix warning about unused codec_dai. * Use devm_kzalloc to manage memory allocation. --- sound/soc/tegra/Kconfig | 9 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 233 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 244 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/tegra_alc5632.c diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index c6af1fd..ce1b773 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -47,3 +47,12 @@ config SND_SOC_TEGRA_TRIMSLICE help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. + +config SND_SOC_TEGRA_ALC5632 + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4d943b3..8e584b8 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -14,6 +14,8 @@ obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o +snd-soc-tegra-alc5632-objs := tegra_alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o +obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c new file mode 100644 index 0000000..89697c6 --- /dev/null +++ b/sound/soc/tegra/tegra_alc5632.c @@ -0,0 +1,233 @@ +/* +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Marc Dietrich <marvin24(a)gmx.de> +* +* 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 <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/alc5632.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-alc5632" + +struct tegra_alc5632 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_alc5632_asoc_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->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 64000: + case 88200: + case 96000: + mclk = 128 * srate; + break; + default: + mclk = 512 * srate; + break; + } + /* FIXME: Codec only requires >= 3MHz if OSR==0 */ + while (mclk < 6000000) + mclk *= 2; + + err = tegra_asoc_utils_set_rate(&alc5632->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_alc5632_asoc_ops = { + .hw_params = tegra_alc5632_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_alc5632_hs_jack; + +static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_alc5632_hs_jack_gpio = { + .name = "Headset Detect", + .report = SND_JACK_HEADSET, + .debounce_time = 150, +}; + +static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { + /* Internal Speaker */ + {"Int Spk", NULL, "SPKOUT"}, + {"Int Spk", NULL, "SPKOUTN"}, + + /* Headset Mic */ + {"MIC1", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Headset Mic"}, + + /* Headset Stereophone */ + {"Headset Stereophone", NULL, "HPR"}, + {"Headset Stereophone", NULL, "HPL"}, +}; + +static const struct snd_kcontrol_new tegra_alc5632_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + +static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack); + snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins), + tegra_alc5632_hs_jack_pins); + snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, + 1, &tegra_alc5632_hs_jack_gpio); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static struct snd_soc_dai_link tegra_alc5632_dai = { + .name = "ALC5632", + .stream_name = "ALC5632 PCM", + .codec_name = "alc5632.0-001e", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "alc5632-hifi", + .init = tegra_alc5632_asoc_init, + .ops = &tegra_alc5632_asoc_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_alc5632 = { + .name = "tegra-alc5632", + .dai_link = &tegra_alc5632_dai, + .num_links = 1, + .controls = tegra_alc5632_controls, + .num_controls = ARRAY_SIZE(tegra_alc5632_controls), + .dapm_widgets = tegra_alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_alc5632_dapm_widgets), + .dapm_routes = tegra_alc5632_audio_map, + .num_dapm_routes = ARRAY_SIZE(tegra_alc5632_audio_map), + .fully_routed = true, +}; + +static __devinit int tegra_alc5632_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_alc5632; + struct tegra_alc5632 *alc5632; + int ret; + + alc5632 = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_alc5632), GFP_KERNEL); + if (!alc5632) { + dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + return ret; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, alc5632); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + tegra_asoc_utils_fini(&alc5632->util_data); + return ret; + } + + return 0; +} + +static int __devexit tegra_alc5632_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&alc5632->util_data); + + return 0; +} + +static struct platform_driver tegra_alc5632_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tegra_alc5632_probe, + .remove = __devexit_p(tegra_alc5632_remove), +}; +module_platform_driver(tegra_alc5632_driver); + +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.3.4

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Re: [alsa-devel] [PATCH v4 4/6] core: add API header and driver header files
by Nallasellan, Singaravelan 13 Dec '11

13 Dec '11

> +struct snd_compr_runtime { > + snd_pcm_state_t state; > + struct snd_compr_ops *ops; > + void *buffer; Can we define buffer as char *? > + size_t buffer_size; > + size_t fragment_size; Can we define buffer_size and fragment_size as unsigned items? Can we have negative size here? > + unsigned int fragments; > + size_t hw_pointer; > + size_t app_pointer; > + size_t bytes_written; > + size_t bytes_copied; > + wait_queue_head_t sleep; > +}; > +

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[alsa-devel] [PATCH v2] ALSA: hda: fix mute led polarity for HP laptops with buggy BIOS
by Gustavo Maciel Dias Vieira 13 Dec '11

13 Dec '11

Some HP laptop models do not have a properly filled OEM string used to set the gpio and polarity of the mute led. Make the mute led configuration work for this case. Signed-off-by: Gustavo Maciel Dias Vieira <gustavo(a)sagui.org> --- Em Seg, 2011-12-12 às 17:10 +0100, Takashi Iwai escreveu: > > Yes, subsytem_id can be used. OTOH, I guess HP_Mute_LED_P_G check > would be easier and may cover more machines. So, I prefer the > latter. > Is this ok? Instead of setting polarity to 1, I can select it by subsystem_id list. Abraços, Gustavo sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index eeb25d52..616678f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4929,6 +4929,12 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity) set_hp_led_gpio(codec); return 1; } + /* BIOS bug: unfilled OEM string */ + if (strstr(dev->name, "HP_Mute_LED_P_G")) { + set_hp_led_gpio(codec); + spec->gpio_led_polarity = 1; + return 1; + } } /* -- 1.7.7.3

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Re: [alsa-devel] ASoC updates for 3.3
by Takashi Iwai 13 Dec '11

13 Dec '11

At Tue, 13 Dec 2011 15:12:54 +0800, Mark Brown wrote: > > The following changes since commit 102477b6380978b1532e5d1f769155026a332aff: > > Merge branch 'for-3.2' into for-3.3 (2011-12-06 10:09:55 +0000) > > are available in the git repository at: > > git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 > > There's a merge up of 3.2-rc3 in here so the diffstat and log are bigger > than they might be otherwise - this is due to the cross tree merges with > the WM8994 MFD side which allow us to eliminate the ASoC version of the > WM8994 register access and cache code, using the much more efficient > regmap version instead. Thanks, pulled now. Though, I've got a few new compile warnings. sound/soc/codecs/wm8903.c:2036:2: warning: initialization discards ‘const’ qualifier from pointer target type sound/soc/codecs/wm9090.c:669:22: warning: unused variable ‘wm9090’ Takashi

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[alsa-devel] [PATCH] ASoC: OMAP: Kconfig string cleanup
by Peter Ujfalusi 13 Dec '11

13 Dec '11

User visible string cleanup in the config file. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/Kconfig | 30 +++++++++++++++--------------- 1 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index fb1bf25..9329810 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,5 +1,5 @@ config SND_OMAP_SOC - tristate "SoC Audio for the Texas Instruments OMAP chips" + tristate "SoC Audio for Texas Instruments OMAP based boards" depends on ARCH_OMAP config SND_OMAP_SOC_DMIC @@ -16,7 +16,7 @@ config SND_OMAP_SOC_HDMI tristate config SND_OMAP_SOC_N810 - tristate "SoC Audio support for Nokia N810" + tristate "Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C depends on OMAP_MUX select SND_OMAP_SOC_MCBSP @@ -25,7 +25,7 @@ config SND_OMAP_SOC_N810 Say Y if you want to add support for SoC audio on Nokia N810. config SND_OMAP_SOC_RX51 - tristate "SoC Audio support for Nokia RX-51" + tristate "Audio support for Nokia RX-51" depends on SND_OMAP_SOC && MACH_NOKIA_RX51 select OMAP_MCBSP select SND_OMAP_SOC_MCBSP @@ -36,7 +36,7 @@ config SND_OMAP_SOC_RX51 hardware. This is also known as Nokia N900 product. config SND_OMAP_SOC_AMS_DELTA - tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + tristate "Audio support for Amstrad E3 (Delta) videophone" depends on SND_OMAP_SOC && MACH_AMS_DELTA select SND_OMAP_SOC_MCBSP select SND_SOC_CX20442 @@ -55,7 +55,7 @@ config SND_OMAP_SOC_AMS_DELTA RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" config SND_OMAP_SOC_OSK5912 - tristate "SoC Audio support for omap osk5912" + tristate "Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC23 @@ -63,7 +63,7 @@ config SND_OMAP_SOC_OSK5912 Say Y if you want to add support for SoC audio on osk5912. config SND_OMAP_SOC_OVERO - tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35" + tristate "Audio support for Gumstix Overo and CompuLab CM-T35" depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -72,7 +72,7 @@ config SND_OMAP_SOC_OVERO Gumstix Overo or CompuLab CM-T35 config SND_OMAP_SOC_OMAP3EVM - tristate "SoC Audio support for OMAP3EVM board" + tristate "Audio support for OMAP3EVM board" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -80,7 +80,7 @@ config SND_OMAP_SOC_OMAP3EVM Say Y if you want to add support for SoC audio on the omap3evm board. config SND_OMAP_SOC_AM3517EVM - tristate "SoC Audio support for OMAP3517 / AM3517 EVM" + tristate "Audio support for OMAP3517 / AM3517 EVM" depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC23 @@ -89,7 +89,7 @@ config SND_OMAP_SOC_AM3517EVM EVM. config SND_OMAP_SOC_SDP3430 - tristate "SoC Audio support for Texas Instruments SDP3430" + tristate "Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -98,7 +98,7 @@ config SND_OMAP_SOC_SDP3430 SDP3430. config SND_OMAP_SOC_SDP4430 - tristate "SoC Audio support for Texas Instruments SDP4430" + tristate "Audio support for Texas Instruments SDP4430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM @@ -109,7 +109,7 @@ config SND_OMAP_SOC_SDP4430 SDP4430. config SND_OMAP_SOC_OMAP4_HDMI - tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" + tristate "Audio support for Texas Instruments OMAP4 HDMI" depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4 select SND_OMAP_SOC_HDMI help @@ -117,7 +117,7 @@ config SND_OMAP_SOC_OMAP4_HDMI OMAP4 chips config SND_OMAP_SOC_OMAP3_PANDORA - tristate "SoC Audio support for OMAP3 Pandora" + tristate "Audio support for OMAP3 Pandora" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -125,7 +125,7 @@ config SND_OMAP_SOC_OMAP3_PANDORA Say Y if you want to add support for SoC audio on the OMAP3 Pandora. config SND_OMAP_SOC_OMAP3_BEAGLE - tristate "SoC Audio support for OMAP3 Beagle and Devkit8000" + tristate "Audio support for OMAP3 Beagle and Devkit8000" depends on TWL4030_CORE && SND_OMAP_SOC depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000) select SND_OMAP_SOC_MCBSP @@ -135,7 +135,7 @@ config SND_OMAP_SOC_OMAP3_BEAGLE the clone Devkit8000. config SND_OMAP_SOC_ZOOM2 - tristate "SoC Audio support for Zoom2" + tristate "Audio support for Zoom2" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -143,7 +143,7 @@ config SND_OMAP_SOC_ZOOM2 Say Y if you want to add support for Soc audio on Zoom2 board. config SND_OMAP_SOC_IGEP0020 - tristate "SoC Audio support for IGEP v2" + tristate "Audio support for IGEP v2" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020 select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 -- 1.7.8

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