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Re: [alsa-devel] Audio support for TLV320AIC23 on Simtec S3C24XX products
by Mark Brown 13 Aug '09

13 Aug '09

On Thu, Aug 13, 2009 at 02:24:30PM +0100, Ben Dooks wrote: > Mark Brown wrote: > >Also, as I've mentioned to you on a number of previous occasions please > >do try to remember to CC subsystem maintainers when submitting patches. > I thought you where the ASoC maintainer? Who else should I be CC'ing? You'd only sent the patches to the list, there wasn't a CC to me. > >So I'm guessing that this is muting through powering off? In that case > >(and probably in general) this would be better implemented as a DAPM > >widget with a SND_SOC_DAPM_PIN_SWITCH() control set up to provide the > >mute functionality. Doing this will allow the speaker to be powered off > >when no audio is being played, saving power and helping avoid any pops > >or clicks being amplified by it. > I tried it that way first, unfortunately the AMP itself makes a POP > when powering up and down, the headphone output of the TLV320AIC23 > doesn't seem to have a problem with pops. But it doesn't pop when people use the mute control? OK, anyway.

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Re: [alsa-devel] tlv320aic3x: Allow driver to use already registered i2c device
by Mark Brown 13 Aug '09

13 Aug '09

On Thu, Aug 13, 2009 at 02:26:38PM +0100, Ben Dooks wrote: > Mark Brown wrote: > >Why would you want to do this? I suspect that perhaps you want to > >convert the driver to use the standard device model instead but without > >understanding your motivation for making this change it is very hard to > >comment in detail. > The device is registered by the code that sets the initial devices > up that have been passed from the bootloader, thus the tlv320aic33 > fails to initialise as there is already an i2c device on the i2c bus > when it comes along. OK, that's what I thought. You need to convert the driver to use the normal device model probe methods rather than using bodges like this. See drivers like wm8731 for an example of how the conversion looks.

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Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

At Thu, 13 Aug 2009 12:41:08 +0300, Mantas Pilipavičius wrote: > > I waited a bit longer and it worked with this command: > speaker-test -c2 > > But again, when I try to play my mp3 files with mp3blaster it doesn't affect > the sound when I use alsamixer. But with speaker-test it does work. It depends how app accesses the sound device. If the application uses "default" PCM, not "hw" or so, then it should work without changing. Or, if app uses OSS emulation over /dev/dsp, softvol doesn't work. Takashi

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[alsa-devel] [PATCH 0/3] patch sequence adds analog devices ad1836 asoc support
by Barry Song 13 Aug '09

13 Aug '09

Hi Mark, The patch sequence adds ad1836 support based on asoc patch 1: new ad1836 codec driver based on asoc patch 2: add set_tdm_slot entry in tdm-dai to define the relationship between audio channel and tdm slot number patch 3: new board driver to connect ad1836 codec and bf5xx patch 3 depends on patch 2. Thanks Barry

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Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

[Please don't drop Cc to alsa-devel ML] At Thu, 13 Aug 2009 11:27:29 +0300, Mantas Pilipavičius wrote: > > So I can play a sound with this command: > speaker-test -Dhw:0,0 -c2 -twav > > I edit my asound.rc: > nano -w ~/.asoundrc and nano -w /etc/asound.conf > > With this: > > pcm.softvol { > type softvol > slave { > pcm "hw:0,0" > } > control { > name "Garsas" > card 0 > } > } > > And the I try: > > speaker-test -Dsoftwol -c2 -twav > > And it doesn't work. Elaborate what doesn't work how. > Sorry to disturb you and consume your precious time, but > I am new to open source and really need this. What about if you run without -Dxxx option? Usually PCM volume should be created as a softvol if not exists. Takashi > On Thu, Aug 13, 2009 at 11:14 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 13 Aug 2009 10:55:38 +0300, > Mantas Pilipavičius wrote: > > > > Can you explain more detailed? What is data manipulation? > > You need to change all the stream data multiplying the volume > gain/attenuation in software because you hardware gives no volume > control by itself. > > In the case of alsa-lib, it's softvol plugin. Or, if you use PA, > PA does it automatically. > > Takashi > > > On Thu, Aug 13, 2009 at 8:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Thu, 13 Aug 2009 00:39:48 +0300, > > Mantas Pilipavičius wrote: > > > > > > http://www.alsa-project.org/db/?f= > > afa01dd48ed59d895d18407af62dbd1fa138686c > > > Everything works great, I can play the sound, but PCM and Master > > channels > > > don't work, so I can't increase or decrease volume. Only thing I > can do > > is > > > mute/unmute PCM channel > > > > This hardware has no volume control indeed. > > The volume has to be adjusted in software by data manipulation. > > > > Takashi > > > > > >

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[alsa-devel] ALSA and MP3
by gammay 13 Aug '09

13 Aug '09

Hello, I am looking at making this product use case possible: Audio samples are streamed over wire/wireless from a source device to a sink device. The streaming format is in mp3. On the source device I need to capture locally played mp3 samples and stream to the sink device. The sink device takes care of uncompressing and playback. Can we capture compressed mp3 streams directly using ALSA? It is possible to capture uncompressed PCM samples using file plug or by other means in ALSA. While it is possible to directly read from the mp3 file, I would rather grap the samples that a media player sends out because a) it takes care of timing b) I can control the media player to play/pause the stream and for playlist management. For the user it is transperent - he/she will use only the local media player to control the streaming. I am not sure if this lies in the domain of ALSA at all - as I understand once we are in ALSA we are dealing with only PCM. In such case, please give me some pointers where I can look at to make this possible. I am using mplayer, but can also look at other types (non-GUI, command-line based). Can we do this in a non-media player specific way? Thank you, Girish

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Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

At Thu, 13 Aug 2009 10:55:38 +0300, Mantas Pilipavičius wrote: > > Can you explain more detailed? What is data manipulation? You need to change all the stream data multiplying the volume gain/attenuation in software because you hardware gives no volume control by itself. In the case of alsa-lib, it's softvol plugin. Or, if you use PA, PA does it automatically. Takashi > On Thu, Aug 13, 2009 at 8:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 13 Aug 2009 00:39:48 +0300, > Mantas Pilipavičius wrote: > > > > http://www.alsa-project.org/db/?f= > afa01dd48ed59d895d18407af62dbd1fa138686c > > Everything works great, I can play the sound, but PCM and Master > channels > > don't work, so I can't increase or decrease volume. Only thing I can do > is > > mute/unmute PCM channel > > This hardware has no volume control indeed. > The volume has to be adjusted in software by data manipulation. > > Takashi > >

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[alsa-devel] [PATCH] new board driver to connect bfin-5xx with ad1836 codec
by Barry Song 13 Aug '09

13 Aug '09

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 8 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 139 +++++++++++++++++++++++++++++++++++++ 3 files changed, 149 insertions(+), 0 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1836.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 8a4de4d..ac927ff 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -88,6 +88,14 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_AD1980 tristate "SoC AD1980/1 Audio support for BF5xx" depends on SND_BF5XX_AC97 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index f4d7607..87e3042 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -16,11 +16,13 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support +snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad1938-objs := bf5xx-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c new file mode 100644 index 0000000..c0ef965 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -0,0 +1,139 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad1836.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 4 2009 + * Description: Board driver for ad1836 sound chip + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad1836.h" +#include "bf5xx-sport.h" + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" + +static struct snd_soc_card bf5xx_ad1836; + +static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad1836_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 *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret = 0; + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI slots, 8 channels, mask is defined as slot seq + * 0x73625140 means audio channel 0 uses slot 0 + * channel 1 uses slot 4 + * channel 2 uses slot 1 + * channel 3 uses slot 5 + * ... + */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x73625140, 8, 32); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops bf5xx_ad1836_ops = { + .startup = bf5xx_ad1836_startup, + .hw_params = bf5xx_ad1836_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai = &bf5xx_tdm_dai, + .codec_dai = &ad1836_dai, + .ops = &bf5xx_ad1836_ops, +}; + +static struct snd_soc_card bf5xx_ad1836 = { + .name = "bf5xx_ad1836", + .platform = &bf5xx_tdm_soc_platform, + .dai_link = &bf5xx_ad1836_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad1836_snd_devdata = { + .card = &bf5xx_ad1836, + .codec_dev = &soc_codec_dev_ad1836, +}; + +static struct platform_device *bfxx_ad1836_snd_device; + +static int __init bf5xx_ad1836_init(void) +{ + int ret; + + bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad1836_snd_device) + return -ENOMEM; + + platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata); + bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev; + ret = platform_device_add(bfxx_ad1836_snd_device); + + if (ret) + platform_device_put(bfxx_ad1836_snd_device); + + return ret; +} + +static void __exit bf5xx_ad1836_exit(void) +{ + platform_device_unregister(bfxx_ad1836_snd_device); +} + +module_init(bf5xx_ad1836_init); +module_exit(bf5xx_ad1836_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

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[alsa-devel] ASoC update for 2.6.32
by Mark Brown 13 Aug '09

13 Aug '09

The following changes since commit e0026beac0f1a6253488478b3f022ff243b012b9: Mark Brown (1): ASoC: Update WM9081 for tdm_slot() API change are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Mark Brown (1): ASoC: Update AD1938 for new TDM slot API sound/soc/codecs/ad1938.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

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[alsa-devel] Intel card master and PCM channels doesn't work
by Mantas Pilipavičius 13 Aug '09

13 Aug '09

http://www.alsa-project.org/db/?f=afa01dd48ed59d895d18407af62dbd1fa138686c Everything works great, I can play the sound, but PCM and Master channels don't work, so I can't increase or decrease volume. Only thing I can do is mute/unmute PCM channel

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