From mboxrd@z Thu Jan 1 00:00:00 1970 From: Carsten Koch Subject: Re: optical SPDIF output on Abit NF7 nforce2 main board. Date: Mon, 01 Sep 2003 19:57:39 +0200 Sender: alsa-devel-admin@lists.sourceforge.net Message-ID: <3F538893.1010801@icem.com> References: <3F532D5E.6040300@icem.com> <3F534D6A.5030706@icem.com> <3F535A42.6020403@icem.com> <3F536C97.5080008@icem.com> Mime-Version: 1.0 Content-Type: text/plain; charset=us-ascii; format=flowed Content-Transfer-Encoding: 7bit Return-path: In-Reply-To: Errors-To: alsa-devel-admin@lists.sourceforge.net List-Help: List-Post: List-Subscribe: , List-Unsubscribe: , List-Archive: To: alsa-devel@lists.sourceforge.net List-Id: alsa-devel@alsa-project.org Takashi Iwai wrote: ... >>Unfortunately, the sound is playing too fast with plughw:0,1 as well. > > > please check /proc/asound/card1/pcm0p/sub0/hw_params during playback. vdr:~ # head /proc/asound/card?/pcm?p/sub?/hw_params ==> /proc/asound/card0/pcm0p/sub0/hw_params <== closed ==> /proc/asound/card0/pcm1p/sub0/hw_params <== access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 44100 (44100/1) period_size: 1024 buffer_size: 8192 tick_time: 10000 > if it shows 44100, perhaps spdif configuration doesn't match with the > request one. please check the spdif status in > /proc/asound/card/ac97#0, too. vdr:~ # head -99 /proc/asound/card?/ac97#0 0-0/0: Realtek ALC650 rev 0 Capabilities : DAC resolution : 20-bit ADC resolution : 18-bit 3D enhancement : Realtek 3D Stereo Enhancement Current setup Mic gain : +0dB [+0dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Extended ID : codec=0 rev=1 LDAC SDAC CDAC DSA=0 SPDIF DRA VRA Extended status : SPCV LDAC SDAC CDAC SPDIF=7/8 SPDIF VRA PCM front DAC : 44100Hz PCM Surr DAC : 44100Hz PCM LFE DAC : 44100Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Copyright Category=0x2 Generation=1 Rate=44.1kHz > you can overwrite the default pcm in ~/.asoundrc with '!' prefix. > for example, > > pcm.!default "hw:0,1" That syntax did not work. It gives me the error message: ALSA lib pcm.c:1787:(snd_pcm_open_conf) Invalid type for PCM default definition (id: default, value: hw:0,1) I changed it to pcm.!default { type hw card 0 device 1 } Should that have the same effect? At least I can now omit the -d parameter for alsaplayer and the sound is still coming out of the SPDIF output (still too fast, though, see above). I can also control the volume via alsaplayer's slider. However, the alsamixer volume control has no effect and kde desktop sounds (i.e. typing ^G) do not work. As I am not really using the analog output, it would be fine with me to simply delete/deactive/hide the analog output PCM device, so the SPDIF output becomes the only PCM device and everything (including the OSS emulation) works with it. Is there a way to do that? Carsten. P.S.: I am still trying to update to the latest CVS, but I keep getting the error message: cvs [update aborted]: end of file from server (consult above messages if any) I will keep trying and report any news on the loop-through as soon as I have successfully installed the latest CVS version. ------------------------------------------------------- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf