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[alsa-devel] Alsa Problem
by prabhu 02 May '08

02 May '08

Respected Sir............ My name is prabhu working in one of the software company in india. when i was trying to use arecord and aplay command in my console(fedora-2.6.11-1.1369_FC4) i am getting following report.... So please kindly describe me for solve this problem...i am newbie in linux... [root@localhost cards]# arecord ALSA lib conf.c:1578:(snd_config_load1) _toplevel_:56:23:No such file or directory ALSA lib conf.c:2821:(snd_config_hook_load) /etc/alsa/cards/aliases.conf may be old or corrupted: consider to remove or fix it ALSA lib conf.c:2684:(snd_config_hooks_call) function snd_config_hook_load returned error: No such file or directory ALSA lib pcm.c:1959:(snd_pcm_open_conf) Invalid type for PCM default definition (id: default, value: cards.pcm.default) arecord: main:564: audio open error: Invalid argument [root@localhost cards]# Thanks &b Regards Prabhu P

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[alsa-devel] help with idt 92hd206 codec for intel hda?
by Greg Zornetzer 02 May '08

02 May '08

Hi all, I recently acquired an ECS GF7100PVT-M3 motherboard, which appears to use the IDT 92HD206 codec for HDA. Unfortunately, it doesn't appear to have full support under ALSA. It gets detected as another Sigmatel chip. Analog stereo output appears to work, but the mixer doesn't find any of the normal inputs. I found the tech specs for the chip at http://www.idt.com/products/getDoc.cfm?docID=13461840 I've attached an lspci output and the codec file to this email. I'm not much of an driver hacker, but if someone could give me an idea of what needs to go into the patch_sigmatel.c file to make this thing work, I'd be willing to try. I'm also more than willing to test out patches on my setup. Thanks in advance for your help. -Greg

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[alsa-devel] When turning up the volume, the sound dies and I'm forced to reboot...
by Sean Thayne 02 May '08

02 May '08

Hi All, I've got a via8237 audio card, and I'm using the latest alsa driver via82xx, and I got a problem. Basically if I start playing say a music file, and then turn the volume up above 75%, My sound craps out and I get a muffled, very low volume version of the song, the only way I've found to fix this is to reboot. This also happens randomly at low volume. I thought it might be a hardware issue, but I've tried it on 3 other boxes with the same card. Thanks for any input at all... -- Sean Thayne

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[alsa-devel] How to support several sound cards using same driver?
by Cliff Cai 01 May '08

01 May '08

Hi all, I have a sound card which has 8 DAC/ADC pairs,and each DAC/ADC pair can be seen as an independent sub-card.An 8-slot buffer struct is introduced to feed/get data to/from one of the 8 sub-cards.For example,if an application wants to play muic using No.7 sub-card,the driver only need to copy the pcm data from PCM layer to No.7 slot and fill the other 7 slots with zero.The driver has been implemented. The problem is how to expose the 8 sub-cards to application developers,so they can see 8 independent device nodesunder /dev?They should be /dev/dsp0,/dev/dsp1,.../dev/dsp7 and card0,card1,...card7.Also when a sub-card is opened(i.e. /dev/dsp7),the driver must know the sub-card's index ,here is 7,so it can choose a relevant slot to feed/get data. Any suggestion is appreciated! Best Regards Cliff _________________________________________________________________ 新年换新颜，快来妆扮自己的MSN给心仪的TA一个惊喜！ http://im.live.cn/emoticons/?ID=18

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[alsa-devel] How to support several cards using the same driver?
by Cliff Cai 01 May '08

01 May '08

Hi all, I have a sound card which has 8 DAC/ADC pairs,and each DAC/ADC pair can be seen as an independent sub-card. An 8-slot buffer struct is introduced to feed/get data to/from the 8 sub-cards.For example, if an application wants to play music using No.x(x,from 0 to 7) sub-card,the driver only need to copy the data from pcm layer to NO.x slot in the buffer struct and fill the other 7 slots with zero. The driver has been implemented now. The problem is how to expose the 8 sub-cards to application,So the application developer can see 8 independant device nodes under /dev.Those are /dev/dsp0,/dev/dsp1,...dev/dsp7(for oss emulation) or card0,card1...card7(for alsa).When one sub-card is opened the driver need to get the index of the sub-card,so it can feed/get data to/from the relevant slot.For example ,if an application opens /dev/dsp7, then the index "7" can be seen by the driver,and the driver know that No.7 slot will be used. Have I made myself clear?Any suggestion is appreciated! Best Regards Cliff . _________________________________________________________________ 多个邮箱同步管理，live mail客户端万人抢用中 http://get.live.cn/product/mail.html

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[PATCH] Include shuffle in 2.6.25 kernels.
by Robert Jarzmik 01 May '08

01 May '08

Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> --- sound/soc/pxa/pxa2xx-ac97.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 86fb9c2..04156d6 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -28,6 +28,7 @@ #include <linux/mutex.h> #include <asm/hardware.h> #include <asm/arch/pxa-regs.h> +#include <asm/arch/pxa2xx-gpio.h> #include <asm/arch/audio.h> #include "pxa2xx-pcm.h" -- 1.5.5.3 --=-=-= Content-Type: text/x-diff Content-Disposition: inline; filename=0005-Revert-wm9713-Expose-the-PLL-configuration.patch

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[alsa-devel] Problem with aloop
by Gustavo da Silva Serra 30 Apr '08

30 Apr '08

Hi, I am developing an application that captures sound and send it through a network, just like a sound server. To test it I am using kaffeine with alsa output plugin. After some time playing continuously, or if I keep moving the position slider backward and forward, I get choppy sound. If I close and reopen the capture stream opened by my sound server, the sound is restored to good quality. Sometimes I have to reopen many times before I get a good sound. When I use kaffeine oss output plugin I don't have this issue. I don't have an audio sound card, so I don't know if this is an aloop issue. Here are some details: - RedHat EL 4 kernel 2.6.9.55-ELsmp 64bit - Alsa 1.0.16 - Xinelib 1.1.12 Please, if more details are needed I will be happy to provide them. Thanks in advance.

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[alsa-devel] [PATCH 1/4] soc - neo1973_wm8753.c cleanup checkpatch issues
by Mark Brown 30 Apr '08

30 Apr '08

From: Graeme Gregory <graeme(a)openmoko.org> Clean up a few issues with the file that checkpatch noted, no functionality changes. Signed-off-by: Graeme Gregory <graeme(a)openmoko.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/neo1973_wm8753.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 962cc20..e2339b9 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -33,7 +33,7 @@ #include <asm/arch/regs-gpio.h> #include <asm/hardware.h> #include <asm/arch/audio.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/arch/spi-gpio.h> #include <asm/plat-s3c24xx/regs-iis.h> @@ -122,7 +122,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, /* set MCLK division for sample rate */ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS ); + S3C2410_IISMOD_32FS); if (ret < 0) return ret; @@ -133,7 +133,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, /* set prescaler division for sample rate */ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4,4)); + S3C24XX_PRESCALE(4, 4)); if (ret < 0) return ret; @@ -222,7 +222,7 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static int neo1973_scenario = 0; +static int neo1973_scenario; static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -233,7 +233,7 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { - switch(neo1973_scenario) { + switch (neo1973_scenario) { case NEO_AUDIO_OFF: snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); @@ -334,7 +334,7 @@ static void lm4857_write_regs(void) static int lm4857_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - int reg=kcontrol->private_value & 0xFF; + int reg = kcontrol->private_value & 0xFF; int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; @@ -349,11 +349,11 @@ static int lm4857_set_reg(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; - if (((lm4857_regs[reg] >> shift ) & mask) == + if (((lm4857_regs[reg] >> shift) & mask) == ucontrol->value.integer.value[0]) return 0; - lm4857_regs[reg] &= ~ (mask << shift); + lm4857_regs[reg] &= ~(mask << shift); lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; lm4857_write_regs(); return 1; @@ -398,7 +398,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { /* example machine audio_mapnections */ -static const char* audio_map[][3] = { +static const char *audio_map[][3] = { /* Connections to the lm4857 amp */ {"Audio Out", NULL, "LOUT1"}, @@ -450,7 +450,7 @@ static const char *neo_scenarios[] = { }; static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios),neo_scenarios), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), }; static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { @@ -521,8 +521,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) /* * BT Codec DAI */ -static struct snd_soc_cpu_dai bt_dai = -{ .name = "Bluetooth", +static struct snd_soc_cpu_dai bt_dai = { + .name = "Bluetooth", .id = 0, .type = SND_SOC_DAI_PCM, .playback = { -- 1.5.5.1

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[alsa-devel] [PATCH] asoc tlv320aic3x: set power bits correctly
by Daniel Mack 30 Apr '08

30 Apr '08

Set power bits in output stage control registers to reflect the "!muted" bits. The codec needs them set in order to operate correctly. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

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[alsa-devel] Patch for alsa-info.sh to include stderr of some tools
by Kristoffer Janke 30 Apr '08

30 Apr '08

Here is a small patch for the alsa-info.sh tool, now the stderr isnt trown away. -- Kristoffer 'spq' Janke 3c3 < SCRIPT_VERSION=0.4.41 --- > SCRIPT_VERSION=0.4.42 80c80 < aplay -l >> $FILE --- > aplay -l >> $FILE 2>&1 84c84 < arecord -l >> $FILE --- > arecord -l >> $FILE 2>&1 105c105 < amixer -c$i>> $FILE --- > amixer -c$i >> $FILE 2>&1

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