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

[alsa-devel] snd-hda-intel: no data when recording from mic, "IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj" logged in kern.log
by Aleksander Adamowski 17 Mar '10

17 Mar '10

Hi! I think I have some interesting problems to analyse for Takashi. Maybe my sound card model is a non typical one, but I have a continuous stream of capture-related problems with it. It's an integrated sound card on my Gigabyte GA-MA74GM-S2H Rev. 1.1 motherboard. "grep ^Codec /proc/asound/card0/codec#0" shows: Codec: Realtek ALC888 I'm attaching the full contents of /proc/asound/card0/codec#0 for identification. But first, some background. With this particular card, I've been continuously hit by the "azx_get_response timeout, switching to single_cmd mode" problem, described here: http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… However, in my case setting probe_mask=1 (experimentally determined) didn't make the problem go away, although it seemed to reduce its frequency (might be just a coincidence). Whenever the dreaded "azx_get_response timeout, switching to single_cmd mode" message appeared in the kernel log, recording stopped working. The card got into the bad state where it would not record sound, and arecord would be giving a stream of "overrun" error messages: arecord -f cd output.wav Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo overrun!!! (at least 27.876 ms long) overrun!!! (at least 26.965 ms long) overrun!!! (at least 16.169 ms long) overrun!!! (at least 11.749 ms long) overrun!!! (at least 11.298 ms long) overrun!!! (at least 11.814 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.747 ms long) overrun!!! (at least 11.819 ms long) overrun!!! (at least 11.149 ms long) overrun!!! (at least 11.257 ms long) overrun!!! (at least 11.315 ms long) overrun!!! (at least 11.651 ms long) overrun!!! (at least 11.745 ms long) overrun!!! (at least 11.235 ms long) overrun!!! (at least 11.169 ms long) overrun!!! (at least 11.776 ms long) overrun!!! (at least 11.391 ms long) overrun!!! (at least 11.669 ms long) overrun!!! (at least 11.714 ms long) overrun!!! (at least 11.671 ms long) overrun!!! (at least 11.190 ms long) What's notable, sound playback would continue to work OK. After being fed up with interrupted Skype conversations (where I weren't immediately aware that the other side stopped hearing me, since I was hearing them), I've decided to upgrade do kernel 2.6.29 (citing the http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… document: "Since 2.6.29 kernel, the driver has a more robust probing method, so this error might happen rarely, though"). After upgrading to 2.6.29 I've removed any extra snd-hda-intel options from modprobe config files. Now the azx_get_response timeout seems to be gone, but I'm experiencing another problem: after some powered up time, recording starts giving empty streams - there's no error, but audio data has zero lengh. e.g. when I arecord, I get a .wav file that is only 44 bytes long (which seems to be the length of the WAV header). When I make a Skype test call to echo123, and it starts playing back my message, it's zero length too - the playback end signal can be heard immediately after the playback start signal. This problem seems to concide with the following error message in the kernel log: hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. Do you have any idea what could be the cause and how to debug this issue further? -- Best Regards, Aleksander Adamowski http://olo.org.pl

2 10

[alsa-devel] [PATCH 3/5] ASoC: DaVinci: CQ93VC Voice Codec
by miguel.aguilar＠ridgerun.com 08 Mar '10

08 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Currently the DM365 is the only SoC that includes this Voice Codec. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 288 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++++ 4 files changed, 323 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..0daca22 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..dff91fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..37f6ba7 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,288 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + snd_soc_add_controls(codec, cq93vc_snd_controls, + ARRAY_SIZE(cq93vc_snd_controls)); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret != 0) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail1; + } + return 0; + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif -- 1.6.0.4

3 3

[alsa-devel] 2.6.30-rc8 skipping regression also affects cmipci
by Frédéric Brière 07 Mar '10

07 Mar '10

Hi everybody, I was only recently able to upgrade from 2.6.26 to 2.6.32 (long story). In so doing, I noticed infrequent occurrences of audio skipping, which I managed to bisect down to c87d973. It would appear that this commit also affects cmipci (CMI8738), albeit not as badly as it did for intel8x0. (I guess I'm the only person left on Earth still using a stand-alone PCI sound card, which is why this was never reported. <g>) Here's a sample log output after turning xrun debugging on: [ 171.555860] PCM: hw_ptr skipping! (pos=6341, delta=16383, period=1024, jdelta=0/85) [ 181.114680] PCM: hw_ptr skipping! (pos=12529, delta=16370, period=1024, jdelta=0/85) [ 181.114770] PCM: hw_ptr skipping! (pos=12534, delta=16375, period=1024, jdelta=0/85) [ 1713.774316] PCM: hw_ptr skipping! (pos=9456, delta=16369, period=1024, jdelta=0/85) [ 1713.774346] PCM: hw_ptr skipping! (pos=9458, delta=16371, period=1024, jdelta=0/85) [ 1713.774409] PCM: hw_ptr skipping! (pos=9461, delta=16374, period=1024, jdelta=0/85) [ 1713.774613] PCM: hw_ptr skipping! (pos=9470, delta=16383, period=1024, jdelta=0/85) [ 1831.092000] PCM: hw_ptr skipping! (pos=1712, delta=16373, period=1024, jdelta=0/85) [ 1953.347453] PCM: hw_ptr skipping! (pos=3723, delta=16316, period=1024, jdelta=0/84) [ 1953.347465] PCM: hw_ptr skipping! (pos=3724, delta=16317, period=1024, jdelta=0/84) [ 1953.347485] PCM: hw_ptr skipping! (pos=3725, delta=16318, period=1024, jdelta=0/84) You'll notice that this occurs much less frequently than it did for intel8x0. However, I left MPlayer on pause for the night, and at some point the kernel started spewing several syslog messages every couple of seconds: [25332.211286] PCM: hw_ptr skipping! [Q] (pos=4, delta=-1073740800, period=1024, jdelta=5/44744/0) [25332.232637] PCM: hw_ptr skipping! [Q] (pos=1029, delta=-1073739771, period=1024, jdelta=6/44749/0) [25332.253957] PCM: hw_ptr skipping! [Q] (pos=2053, delta=-1073738747, period=1024, jdelta=5/44755/0) [25332.275292] PCM: hw_ptr skipping! [Q] (pos=3077, delta=-1073737723, period=1024, jdelta=5/44760/0) [25332.296616] PCM: hw_ptr skipping! [Q] (pos=4101, delta=-1073736699, period=1024, jdelta=6/44765/0) [25332.317932] PCM: hw_ptr skipping! [Q] (pos=5125, delta=-1073735675, period=1024, jdelta=5/44771/0) [25332.339269] PCM: hw_ptr skipping! [Q] (pos=6149, delta=-1073734651, period=1024, jdelta=5/44776/0) [25332.360600] PCM: hw_ptr skipping! [Q] (pos=7173, delta=-1073733627, period=1024, jdelta=6/44781/0) [25332.381942] PCM: hw_ptr skipping! [Q] (pos=8197, delta=-1073732603, period=1024, jdelta=5/44787/0) [25332.403261] PCM: hw_ptr skipping! [Q] (pos=9221, delta=-1073731579, period=1024, jdelta=5/44792/0) [25337.244901] PCM: hw_ptr skipping! [Q] (pos=12293, delta=-1073728507, period=1024, jdelta=6/44808/0) [25337.266223] PCM: hw_ptr skipping! [Q] (pos=13317, delta=-1073727483, period=1024, jdelta=5/44813/0) This kept going on until I quit MPlayer (which, as I said, was not actually playing anything all this time). Not that skipped silence is much of an issue, but I thought this might add useful information. Let me know if I can be of any help in resolving this. -- If you want to travel around the world and be invited to speak at a lot of different places, just write a Unix operating system. -- Linus Torvalds

2 3

Re: [alsa-devel] HD-Audio device unsuppored (Nvidia G2xx-series GPU implemented)
by Stefan Ringel 02 Mar '10

02 Mar '10

I forgot this attachment 1.patch and alsa-info.txt.

4 20

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.31-rc8-407-gfc9e4a3
by Takashi Iwai 26 Feb '10

26 Feb '10

At Tue, 6 Oct 2009 08:36:04 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > commit fc9e4a3811567f8d1d4c263a89e27b52c90d6098 > Author: Clemens Ladisch <clemens(a)ladisch.de> > AuthorDate: Tue Oct 6 08:21:04 2009 +0200 > Commit: Clemens Ladisch <clemens(a)ladisch.de> > CommitDate: Tue Oct 6 08:21:04 2009 +0200 > > sound: via82xx: move DXS volume controls to PCM interface > > The "VIA DXS" controls are actually volume controls that apply to the > four PCM substreams, so we better indicate this connection by moving the > controls to the PCM interface. > > Commit b452e08e73c0e3dbb0be82130217be4b7084299e in 2.6.30 broke the > restoring of these volumes by "amixer restore" that most distributions > use; the renaming in this patch cures that regression by preventing > amixer from applying the old, wrong volume levels to the new controls. > http://bugzilla.kernel.org/show_bug.cgi?id=14151 > http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613 > > Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> > Cc: <stable(a)kernel.org> Well, this control was once removed, and then revived due to complains by users who have used it in mixer. Moving this to PCM iface does almost the same thing, disabling an easy way to access it. I'm afraid that this patch might be regarded as another "regression". For such an old driver, the compatibility is the most important item. If any, we must provide a way to back to the old behavior. thanks, Takashi > > ----------------------------------------------------------------------- > > Diffstat: > sound/pci/via82xx.c | 27 ++++++++++++++++++--------- > 1 files changed, 18 insertions(+), 9 deletions(-) > > > Patch -------------- (max 1500 lines) > diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c > index acfa476..91683a3 100644 > --- a/sound/pci/via82xx.c > +++ b/sound/pci/via82xx.c > @@ -1626,7 +1626,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > { > struct via82xx *chip = snd_kcontrol_chip(kcontrol); > - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); > + unsigned int idx = kcontrol->id.subdevice; > > ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0]; > ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1]; > @@ -1646,7 +1646,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > { > struct via82xx *chip = snd_kcontrol_chip(kcontrol); > - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); > + unsigned int idx = kcontrol->id.subdevice; > unsigned long port = chip->port + 0x10 * idx; > unsigned char val; > int i, change = 0; > @@ -1705,11 +1705,12 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = > }; > > static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { > - .name = "VIA DXS Playback Volume", > - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .iface = SNDRV_CTL_ELEM_IFACE_PCM, > + .device = 0, > + /* .subdevice set later */ > + .name = "PCM Playback Volume", > .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | > SNDRV_CTL_ELEM_ACCESS_TLV_READ), > - .count = 4, > .info = snd_via8233_dxs_volume_info, > .get = snd_via8233_dxs_volume_get, > .put = snd_via8233_dxs_volume_put, > @@ -1936,10 +1937,18 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) > } > else /* Using DXS when PCM emulation is enabled is really weird */ > { > - /* Standalone DXS controls */ > - err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip)); > - if (err < 0) > - return err; > + for (i = 0; i < 4; ++i) { > + struct snd_kcontrol *kctl; > + > + kctl = snd_ctl_new1( > + &snd_via8233_dxs_volume_control, chip); > + if (!kctl) > + return -ENOMEM; > + kctl->id.subdevice = i; > + err = snd_ctl_add(chip->card, kctl); > + if (err < 0) > + return err; > + } > } > } > /* select spdif data slot 10/11 */ > ------- End of patch > > hooks/update > --- > Git Source Code Management System > hooks/update refs/heads/master \ > 62f08ef9739c5cb5fff5b5bdec3bc02d522b9a12 \ > fc9e4a3811567f8d1d4c263a89e27b52c90d6098 > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog >

6 22

[alsa-devel] Getting pcm_usb_stream plugin to know its limits.
by michael＠cadilhac.name 22 Feb '10

22 Feb '10

Hi there everyone, I've been fighting for a few hours with a USB sound device (namely, Tascam US-144), to get it to work with Audacity (this not being the relevant part). At first, Audacity died in a painful segfault, without any explanation, when associated with the usb_stream plugin[1]. Audacity uses PortAudio, which checks for a few possible parameters of audio rate to find the list of accepted ones. It tries to set a parameter (snd_pcm_hw_params), and if it fails, closes the device, and continues its way. Suppose the `prepare` act of the usb_stream is aborted because of wrong parameters. Then, PortAudio tries to close the device, by calling snd_pcm_close. This in turn calls snd_pcm_drop, which calls the fast_op drop (that of ioplug), which is snd_pcm_ioplug_drop, which then calls the *stop* callback of the usb_stream (snd_pcm_us_stop). The stream not being properly initialized, us->uus->write_area is nil, and the memset fails. This is the modification I did for that purpose : Next, Audacity stopped segfaulting but still didn't work: it couldn't manage to configure the plugin properly. Here's how PortAudio searches for an acceptable audio rate. For each possible sample rate, it gets a full hw_params for the device: snd_pcm_hw_params_any( pcm, hwParams ); then set the sample rate to the hwParams. It then tries to set those params to the device: snd_pcm_hw_params( pcm, hwParams ); The period_size parameter is left untouched, and by default, the pcm inherits of a high value (maybe the highest). Now the problem is the following. pcm_usb_stream says that the maximum accepted period_size is 64*4096: (err = snd_pcm_ioplug_set_param_minmax(&us->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 128, 64*4096) BUT the us122l driver[2] (from 2.6.32) explicitly prohibits period_size greater than 0x3000: if ((cfg->sample_rate != 44100 && cfg->sample_rate != 48000 && (!high_speed || (cfg->sample_rate != 88200 && cfg->sample_rate != 96000))) || cfg->frame_size != 6 || cfg->period_frames > 0x3000) err = -EINVAL; This is now what happens in PortAudio: PortAudio tries each and every sound rate with the "default" period_size, this being greater than 0x3000. It follows that no sound rate will be accepted, because of the period_size, and the device fails to initialize. I'm not sure whose fault is this, and for the time being, I just did the following: Well, that's all folks. Happy new year to everyone. Footnotes: [1] Configured as pcm.!usb_stream { type usb_stream card "1" } which works with a fine-tuned aplay. [2] linux/sound/usb/usx2y/us122l.c -- Michaël `Micha' Cadilhac (LITQ, U. de Montréal) -- http://michael.cadilhac.name || La culture, c'est comme l'amour. || Faut y aller à petits coups au début pour bien en jouir plus tard. || -- Pierre Desproges

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[alsa-devel] [PATCH 0/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 15 Feb '10

15 Feb '10

The following patch set adds suspend/resume support to all echoaudio drivers. It has been tested on an IndigoIOx and a Gina3G. Rawmidi device has been tested partially. PCM and controls interfaces have been tested more and I think it's reasonably safe. I also have a Gina24 and a Gina3G, but I didn't test suspend/resume with them because I couldn't manage to make PM test system work on my pmac. Echoaudio soundcards do not support power management and Dx PCI states. Anyway, the card can enter a "comatose" state by sending a command to the DSP that is supposed to turn parts of the hardware off and reduce power consumption. Doing that causes the loss of the card's internal state, thus it is impossible, upon resume, to make the card continue from the exact point where it was suspended. The driver restores all controls and PCM settings and the applications have to call prepare() to restart the substreams. Rawmidi is restarted automatically, but some data in the i/o buffers may be lost. -- Giuliano.

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Re: [alsa-devel] HP 6930p mute button/led out of sync; fixed by suspend/resume cycle
by drago01 12 Feb '10

12 Feb '10

On Sun, Jan 10, 2010 at 3:31 PM, drago01 <drago01(a)gmail.com> wrote: > Hi, > > I experience an odd issue with my HP6930p laptop. > > After boot pressing the mute button/led causes the audio to be muted > but the sate is not reflected in software (i.e the mixer does not show > the control as muted). > However pressing after a suspend/resume cycle it works as expected. > The hardware mute button/led and the software control are in sync > (pressing the hw button toggles the mixer to mute/unmute; muting the > software mixer results into the led turning red). > > This was with 2.6.31.9 and 2.6.32.1; also installing alsa-drivers from > source does not change this. > > It seems something is not initialized correctly during boot up but is > done on resume, any pointers on how to debug this? > (Resending to correct list)

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[alsa-devel] Sound distortion with Realtek ALC889A
by Damien Brossard 11 Feb '10

11 Feb '10

Hi. I recently bought a Gigabyte GA-MA78GM-S2H motherboard. This board uses the (in)famous (from what i've read for now) alc889a chip. While playing 5.1 content, I can hear noise or distortion coming from the center channel. I can not test my configuration with another OS, so I can't fully dismiss a HW problem. However, i made several tests with 5.1 and 2.0 speakers which points me to soundcard problem. I've made some tests with speaker-test (speaker-test -D surround51 -c 6) : 1) Each time a sound is played either on front left channel or rear left channel, a noise can be heared on the center channel (and subwoofer, this is confirmed pluging a stereo headphone on the center/lfe jack). No noise when a sound is played on right channels (front or rear). 2) In alsamixer, if I set the volume to 0 for the front or rear left channels, i can hear the noise on center/lfe when a sound is "played" on this channels 3) In alsamixer, if I mute the front or rear left channels, i can hear no more noise on center/lfe Complementary tests have been made with alsaplayer and using a simple mp3 file. In this configuration (stereo output), center and rear channels play the same sound as front channel. 1) If all channels are unmuted, I hear no noise. 2) If either the front channel or the rear channel are muted, I can hear the noise on center channel. I tried this with alsa-driver 1.0.20 and 1.0.21 with the same result, I recently went back to 1.0.20 due to kernel upgrade. Things that seem strange to me (maybe not so strange to alsa dev :)) : 1) alsamixer shows volumes in percentage, however when I set a channel volume to 50% and the master volume to 50%, I hope to find a resulting volume of 25%, not 0. 2) lspci reports my soundcard subsystem id as 1458:a022, but in /proc/asound/card0/codec#0 the subsystem id is 1458:a002 Googling around made me find few references (in alsa-devel list) to a similar problem : http://mailman.alsa-project.org/pipermail/alsa-devel/2008-July/008942.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-August/009754.html The distortion / noise described here is exactly what i can hear with my system. Apparently the user solved the issue buying another soundcard. I'd like to be sure it is the only way to go. Attached is the file produced by alsa-info.sh Thanks in advance. Best regards Damien Brossard

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[alsa-devel] [PATCH] ALSA: ASoC: cs4270: enable regulators at probe time
by Daniel Mack 09 Feb '10

09 Feb '10

Enable the bulk regulators at probe time so we can safely disable them again when going to suspend without confusing the reference counter. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8b54575..70c5edb 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -617,8 +617,17 @@ static int cs4270_probe(struct platform_device *pdev) if (ret < 0) goto error_free_pcms; + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret < 0) + goto error_free_regulators; + return 0; +error_free_regulators: + regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + error_free_pcms: snd_soc_free_pcms(socdev); @@ -638,6 +647,7 @@ static int cs4270_remove(struct platform_device *pdev) struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return 0; -- 1.6.3.3

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