[linux-audio-dev] Merging digital sampling sessions

David Kastrup dak at gnu.org
Sun Dec 11 22:44:32 UTC 2005


Wolfgang Woehl <tito at rumford.de> writes:

> David Kastrup <dak at gnu.org>:
>
>> arecord -D spdif -f cd
>>
>> Ok, and here is the rub: playback is not perfect, and neither seems
>> to be the sampling.  Probably at times error interpolation sets in,
>> probably at times samples get lost or replicated.  Not much, perhaps
>> once per minute or so.  This leads to audible clicks, of course.
>
> You're assuming a flawed spdif stream and, well, people keep suggesting 
> changing the spdif cable. It's supposed to have an effect on stream 
> quality worth mentioning. Don't know myself.
>
> But try setting the c-media card to respect the incoming spdif clock. 
> That should make a difference.

Any idea how?  I have had just a few weird experiences: without any
preparation, recording (and monitoring) is just one crackly mess.  If
I do
aplay /usr/share/sounds/startup.wav
while arecord -D spdif has not yet given up with an error, (the
recording has this muted), then the recording will turn into working
pretty much transparently.  However, starting a new recording session
will revert to snack, crackle and pop.

If I do
aplay -D spdif /usr/share/sounds/startup.wav
however, a subsequent arecord -D spdif command will not revert to
crackle mode.

So there is already something going on which I don't understand.  In
"crackle mode", there is basically just noise, but the noise level is
correlated with the amplitude of the sound source.

So what would one actually have to do to have the "incoming spdif
clock" respected?

-- 
David Kastrup, Kriemhildstr. 15, 44793 Bochum



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