At Mon, 01 Sep 2003 19:57:39 +0200,
Carsten Koch wrote:
>
> 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
ok, then apparently it looks like working correctly.
can you play mp3 on normal analog output correctly?
> > 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)
yeah, the above was incorrect.
>
> I changed it to
>
> pcm.!default {
> type hw
> card 0
> device 1
> }
>
> Should that have the same effect?
yes.
> 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
because it's digital output.
> and kde desktop sounds
> (i.e. typing ^G) do not work.
perhaps artsd is running on oss mode.
> 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?
so far, only by using oss-wrapper library.
>
> 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.
blame sourceforge :)
Takashi
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