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

[alsa-devel] [PATCH] sst_platform: add hw_free callback
by xingchao 04 May '11

04 May '11

Signed-off-by: xingchao <xingchao.wang(a)intel.com> --- sound/soc/mid-x86/sst_platform.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 9765fb8..5a946b4 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -380,6 +380,11 @@ static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + static struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, .close = sst_platform_close, @@ -388,6 +393,7 @@ static struct snd_pcm_ops sst_platform_ops = { .trigger = sst_platform_pcm_trigger, .pointer = sst_platform_pcm_pointer, .hw_params = sst_platform_pcm_hw_params, + .hw_free = sst_platform_pcm_hw_free, }; static void sst_pcm_free(struct snd_pcm *pcm) -- 1.7.1

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[alsa-devel] [PATCH 0/4] Implement control sharing
by Stephen Warren 03 May '11

03 May '11

Control sharing is enabled when two widgets include pointers to the same kcontrol_new in their definition. Specifically: static const struct snd_kcontrol_new adcinput_mux = SOC_DAPM_ENUM("ADC Input", adcinput_enum); static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = { SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), }; This is useful when a single register bit or field affects multiple muxes at once. The common case is to have separate control bits or fields for each mux (channel). An alternative way of looking at this is that the mux is a stereo (or even n-channel) mux, rather than independant mono muxes. I have tested this on NVIDIA Tegra Seaboard, specifically with the WM8903 "ADC Input" control, and a locally-enabled digital mic setup. I have not compiled nor tested any other platforms. This patchset does not implement control sharing for dapm_new_mixer() yet. I can do so after discussion/review of this patch series, although note that I don't have anything to test the actual sharing on for such controls. Stephen Warren (4): ASoC: s/w->kcontrols/w->kcontrol_news/g ASoC: Add w->kcontrols, and populate it ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol ASoC: Implement mux control sharing include/sound/soc-dapm.h | 75 +++++---- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/soc-dapm.c | 315 ++++++++++++++++++++++++++----------- 3 files changed, 266 insertions(+), 126 deletions(-)

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[alsa-devel] [PATCH] ASoC: JZ4740: Fix i2s shutdown
by Lars-Peter Clausen 03 May '11

03 May '11

The i2s shutdown callback has the check whether it should be disabled reversed. Currently it is disabled if another stream is still active, but kept enabled if the last stream is closed. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 419bf4f..cd22a54 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -133,7 +133,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - if (!dai->active) + if (dai->active) return; conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); -- 1.7.2.5

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[alsa-devel] [PATCH 1/4] ASoC: Free the card's DAPM context
by Lars-Peter Clausen 03 May '11

03 May '11

Free the card's DAPM context when the card is removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a823654..6a3cb53 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2047,6 +2047,8 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) if (card->remove) card->remove(card); + snd_soc_dapm_free(&card->dapm); + kfree(card->rtd); snd_card_free(card->snd_card); return 0; -- 1.7.2.5

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[alsa-devel] [PATCH v3] ASoC: tegra: TrimSlice machine support
by Mike Rapoport 03 May '11

03 May '11

Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- v3 changes: * s/SND_TEGRA_SOC/SND_SOC_TEGRA * drop 'default' setting in Kconfig v2 changes: * address Stephen and Mark comments * rebase against Mark's for-next branch * switch to data based specification of widgets and routes * explicitly disable unused codec pins sound/soc/tegra/Kconfig | 8 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/trimslice.c | 228 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 238 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/trimslice.c diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 14f7119..1e75da3 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -26,3 +26,11 @@ config SND_SOC_TEGRA_WM8903 boards using the WM8093 codec. Currently, the supported boards are Harmony, Ventana, Seaboard, Kaen, and Aebl. +config SND_SOC_TEGRA_TRIMSLICE + tristate "SoC Audio support for TrimSlice board" + depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_TLV320AIC23 + help + Say Y or M here if you want to add support for SoC audio on the + TrimSlice platform. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 13bef8d..fa6574d 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -11,5 +11,7 @@ obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o +snd-soc-tegra-trimslice-objs := trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o +obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c new file mode 100644 index 0000000..8fc07e9 --- /dev/null +++ b/sound/soc/tegra/trimslice.c @@ -0,0 +1,228 @@ +/* + * trimslice.c - TrimSlice machine ASoC driver + * + * Copyright (C) 2011 - CompuLab, Ltd. + * Author: Mike Rapoport <mike(a)compulab.co.il> + * + * Based on code copyright/by: + * Author: Stephen Warren <swarren(a)nvidia.com> + * Copyright (C) 2010-2011 - NVIDIA, Inc. + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/tlv320aic23.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-trimslice" + +struct tegra_trimslice { + struct tegra_asoc_utils_data util_data; +}; + +static int trimslice_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_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 128 * srate; + + err = tegra_asoc_utils_set_rate(&trimslice->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "codec_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "cpu_dai fmt not set\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 trimslice_asoc_ops = { + .hw_params = trimslice_asoc_hw_params, +}; + +static const struct snd_soc_dapm_widget trimslice_dapm_widgets[] = { + SND_SOC_DAPM_HP("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route trimslice_audio_map[] = { + {"Line Out", NULL, "LOUT"}, + {"Line Out", NULL, "ROUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, +}; + +static int trimslice_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_dapm_nc_pin(dapm, "LHPOUT"); + snd_soc_dapm_nc_pin(dapm, "RHPOUT"); + snd_soc_dapm_nc_pin(dapm, "MICIN"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .codec_name = "tlv320aic23-codec.2-001a", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "tlv320aic23-hifi", + .init = trimslice_asoc_init, + .ops = &trimslice_asoc_ops, +}; + +static struct snd_soc_card snd_soc_trimslice = { + .name = "tegra-trimslice", + .dai_link = &trimslice_tlv320aic23_dai, + .num_links = 1, + + .dapm_widgets = trimslice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(trimslice_dapm_widgets), + .dapm_routes = trimslice_audio_map, + .num_dapm_routes = ARRAY_SIZE(trimslice_audio_map), +}; + +static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_trimslice; + struct tegra_trimslice *trimslice; + int ret; + + trimslice = kzalloc(sizeof(struct tegra_trimslice), GFP_KERNEL); + if (!trimslice) { + dev_err(&pdev->dev, "Can't allocate tegra_trimslice\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); + if (ret) + goto err_free_trimslice; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, trimslice); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&trimslice->util_data); +err_free_trimslice: + kfree(trimslice); + return ret; +} + +static int __devexit tegra_snd_trimslice_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&trimslice->util_data); + + kfree(trimslice); + + return 0; +} + +static struct platform_driver tegra_snd_trimslice_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_snd_trimslice_probe, + .remove = __devexit_p(tegra_snd_trimslice_remove), +}; + +static int __init snd_tegra_trimslice_init(void) +{ + return platform_driver_register(&tegra_snd_trimslice_driver); +} +module_init(snd_tegra_trimslice_init); + +static void __exit snd_tegra_trimslice_exit(void) +{ + platform_driver_unregister(&tegra_snd_trimslice_driver); +} +module_exit(snd_tegra_trimslice_exit); + +MODULE_AUTHOR("Mike Rapoport <mike(a)compulab.co.il>"); +MODULE_DESCRIPTION("Trimslice machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.3.1

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[alsa-devel] Copy call back for capture substream
by kshemanth.vasireddi＠wipro.com 03 May '11

03 May '11

Hi, I am new to ALSA, I am trying to write an ALSA driver for voice channels of and ADC. I had create a sound card with a PCM instance with 2 capture sub streams. I want to copy data manually by copy call back. I had registered a copy call back. I am trying to test my driver with arecord -c 1 -f s16_LE -t wav test.wav. In log all call backs are invoked, but copy call back is not getting invoked. Can you please let me know when will copy call back shall be invoked, is there any condition for it. Thanks and regards, Kshemanth Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[alsa-devel] [PATCH 0/10] Realtek auto-parser updates
by Takashi Iwai 02 May '11

02 May '11

Hi, this is a series of patches I'm going to apply as 2.6.40 material. Basically this adds the missing features to BIOS auto-parser mode, the support of multi-channel setup (2/4/6-channels) and the auto-mute of line-out jacks as well as speakers. Many changes have been done but mostly in systematic manner. I've checked through hda-emu and a few machines with Realtek codecs, but of course can't cover all changes I made. If you see any issues with the updated tree, please let me know. thanks, Takashi

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[alsa-devel] SALSA-library version 0.1.0 released
by Takashi Iwai 29 Apr '11

29 Apr '11

Hi, my old small pet project, salsa-lib, was refreshed recently and bumped to version 0.1.0 after a long pause. Now it can build a drop-in libasound.so.* that is ABI-compatible with the real alsa-lib. At least, the programs in alsa-utils (except for sequencer stuff) and PulseAudio daemon worked somehow. The source tarball is found at: ftp://ftp.suse.com/pub/people/tiwai/salsa-lib/salsa-lib-0.1.0.tar.bz2 I converted the local hg tree to git tree and published at: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/salsa-lib.git Takashi

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Re: [alsa-devel] [PATCH] - ALSA hda ad1988 - Add Independent HP Switch to model 6stack-dig-fp
by Takashi Iwai 29 Apr '11

29 Apr '11

At Thu, 28 Apr 2011 21:10:23 +0800, Raymond Yau wrote: > > 2011/4/26 Takashi Iwai <tiwai(a)suse.de> > > > At Tue, 19 Apr 2011 08:12:57 +0800, > > Raymond Yau wrote: > > > > > > Add Independent HP Switch to model 6stack-dig-fp > > > > > > allow user to switch Independent HP on/off by using ALSA specific "share > > > stream" mode > > > > > > This don't need to make "Headphone Playback Volume" inactive but this > > also > > > mean that "Headphone Playback Volume" is no longer slave of the "Virtual > > > Master" any more > > > > The implementation looks almost OK (although there are a few pieces > > of unused code slike activate_ctl()), I prefer an enum control for it. > > > > > The other ad1988 models does not fully utilised five DACs > > This implementation is specific to 10 channels HDA ,as Front and Headphone > just share the stream when "Independent HP" is off > > The "Independent HP" switch have to be write protected when any application > is using multiout or alt_playback , alsamixer just display the inactive > control in grey color. > > The driver return -EBUSY if any application try to open alt_playback when > independent HP is switch off. If it's really implemented :) You defined activate_ctl() but it's called from nowhere. > As for the upmixing feature, I wonder why no one complain about sending left > channel to the center and right channel to subwoofer Well, maybe because usually you set up 5.1 properly when you have such systems, then upmixing right in the backend, not in the driver. > > It's a frequently seen problem that user sets "Independent HP" mixer > > switch accidentally and gets unexpected results. An enum control > > tends to be more clear than a mixer switch in general. > > > > This depend on how the mixer application treat the "inactive state" of > control (e.g. qasmixer make the "inactive" control invisible ) No, this is a question how easily such feature can be changed mistakenly. It's no technical question, but a design issue. The mixer element itself isn't so self-explanatory, and people tend to turn on every mixer switch once when something doesn't work and wonder what's wrong. > Still unable to enable the unsolicited event for jack sense even if I add > the unsol_event and verb for the audio jacks at rear panel since I don't > have the HDA compilant front audio panel > > SoundMax automatically popup immediately when jack is plug into the rear > panel at other OS, so the hardware is capable of jack sense at rear panel > > AFG Function Id: 0x1 (unsol 0) > > Is there any trick to enable/debug the unsolicited event ? > seem unsol event is disabled in the HDA controller No, there shouldn't be big differences. Otherwise it won't work on any other machines. There might be other way to trigger the jack detection, e.g. GPIO unsolicited event, depending on the codec chip. I don't remember how is AD1988, though... Takashi

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[alsa-devel] [PATCH] ALSA: HDA: Fix automute for Gateway NV79
by David Henningsson 29 Apr '11

29 Apr '11

The PCI SSID is 1025:031c and the codec SSID is 1025:031d, so the driver mistakes this for a SKU value, but looking at the numbers, this is obviously wrong. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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