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[alsa-devel] [ALC262 patch_realtek.c] Fujitsu Lifebook PC speaker signal
by Tony Vroon 14 Apr '08

14 Apr '08

Good afternoon, Could you please consider the following patch for the 2.6.26 merge window. The legacy PC speaker signal was not routed to outputs. The codec is not prevented from powering down in this patch, although I suppose one could argue that perhaps it should be. Let me know if anyone feels strongly one way or the other. Signed-Off-By: Tony Vroon <tony(a)linx.net> Regards, Tony V. LINX sysadmin

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[alsa-devel] [PATCH]PCI quirk for laptop LG which use CMI9880
by Jiang zhe 14 Apr '08

14 Apr '08

Hi, Please refer to [0003874] on the alsa mantis. This patch added the pci quirk. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

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[alsa-devel] [PATCH] soc - Include register in DAPM debug output
by Mark Brown 14 Apr '08

14 Apr '08

When logging register changes in DAPM debug output include the register number. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index abac684..e0637d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -226,7 +226,7 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(POP_TIME); } - dbg("reg old %x new %x change %d\n", old, new, change); + dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); return change; } -- 1.5.4.4

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[alsa-devel] [PATCH]PCI quirk for MSI laptop
by Jiang zhe 14 Apr '08

14 Apr '08

Hi, Please refer to [0003848] on the alsa mantis. This patch adds the pci quirk and Mic-Int controller. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

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[alsa-devel] [PATCH] snd_usb_caiaq: use likely()/unlikely()
by Daniel Mack 13 Apr '08

13 Apr '08

Use likely() and unlikely() in code which is called regularily. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

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[alsa-devel] External IO plugin question
by Taek Joo Kim 12 Apr '08

12 Apr '08

Hi. I am working on ALSA external IO plugin now. Its function is similar to the pulse plugin. A server sends sound to multiple clients. When I play sound with my plugin, the sound is played really fast. It means that it is like playing with x5 or x10 speed. I did not write any code for synchronization between server and client. To try to fix it, I saw the pulse plugin. I guess that it uses some synchronization code (poll_descriptors), but I am not sure. My questions are: 1. Which part should take care of playing sound properly in this architecture, server or client? 2. Do I need to deal with synchronization between server and client? 3. If 2 is yes, is there any hints to do this with multiple clients? Thank you in advance.

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[alsa-devel] Softvol for Independent Volume Control
by Cameron Barfield 11 Apr '08

11 Apr '08

Hi -- I've got an application that receives multiple network audio streams. Using DMIX, I am able to play back all the streams through my sound card. The problem is that I would like to be able to independently adjust the volume of each audio stream. Would Softvol work for this? All the examples I've found only show Softvol working as a replacement for a master volume control (and that's all I've been able to get it to do). If Softvol won't work for independent volume controls, is there any other ALSA plugin or API call that I'm missing? Switching to something like Pulseaudio would probably not be an option since I'm running in an embedded environment. Regards, Cameron

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[alsa-devel] XRUN happens too often.
by Rong-Jhe 10 Apr '08

10 Apr '08

I use "default" pcm and open it in block mode. Then, I set the this pcm to perform 16-bit, two channels, and 44100 bps sampling rate audio data. The number of periods per buffer is 10. The follows are the parameters that I read from the pcm. sampling rate = 44100 bps period time = 21333 us period size = 940 frames buffer time = 213330 us buffer size = 9408 frames A sample in 16-bit, two channels is 4 bytes. I allocate a audio data of 9408x4=37632 bytes. audio_data=malloc(buffer_size*4); This audio_data buffer will receive audio data from the network. If this buffer is full, I use snd_pcm_writei( ) to put the buffer into the pcm. snd_pcm_sframes_t frames; snd_pcm_t *handle; while((frame=snd_pcm_writei(handle, audio_data, buffer_size))<0) { if(frames==-EPIPE) snd_pcm_prepare(handle); else printf("snd_pcm_writei error %d\n", frames); } The codes states that I put the entire buffer into the pcm at each time. The pcm use 10 periods to perform this buffer. The program can work normally. If the sampling rate is changed, the program will encounter XRUN frequently. The follows are the parameters that I read from the pcm after setting sampling rate to 48000. sampling rate = 48000 bps period time = 21333 us period size = 1024 frames buffer time = 213330 us buffer size = 10240 frames audio_data=malloc(buffer_size*4); //10240x4=40960 I also put the entire buffer into the pcm at each time, but snd_pcm_writei( ) will return -PIPE frequently. What is the problem?

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[alsa-devel] M-Audio Fast Track Ultra USB
by knoxit.etc＠gmail.com 10 Apr '08

10 Apr '08

Hello, I was wondering if any user had experiences with the M-Audio Fast Track Ultra USB unit and ALSA. I'm not sure if its supported, it doesn't show up on the soundcard matrix. If not, is there any development taking place right now? Thanks.

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[alsa-devel] [PATCH] snd_usb_caiaq: correct input channel order
by Daniel Mack 09 Apr '08

09 Apr '08

This patch corrects the input channel order of hardware supported by snd_usb_caiaq. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

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