Subject: buffering question
From: The Eye (the.eye_AT_gmx.li)
Date: Wed Dec 25 2002 - 18:24:05 EET
I mainly use ecasound for recording-purposes, e.g. recording really long
stuff from the radio and so on (using ecasound because of the largefile
Been using some 2.1.something up untill a few days ago .. pulled the new
CVS, compiled that and use that ..
so since I saw the -B option in the man-page I tried that out
(-B:rtlowlatency) for my last recording session .. but right in the
middle of recording (after like 1.5 hours) I start getting lots of
drop-outs .. and error messages that say:
(audioio-alsa) warning! playback overrun - samples lost! Break was at least 0.21 ms long.
So when thinking about this I realised I don't know much about buffering
and what it all means .. I mean the man-page says I can use
-B:auto|nonrt|rt|rtlowlatency ... plus there is this -z:db feature and
the fact that I can use -b:buffer_size ...
at some stage (with -b I think) there is mention of "for real-time
processing you should set this as low as possible" ..
Now I'm wondering .. what _is_ real-time processing as applied to a
tool like ecasound? and what is it that I am doing (recording long
stretches of audio, preferably so that any other acitivity on my PC will
have no negative effect on my recording), and what would be good
settings for that? I even realised that I don't really understand what
"low latency" actually means.
So if anyone can shed some light on this or point me in the direction of
some howto/faq/explaining documents, I'd be real happy.
Hardware is (dunno if this is important):
Athlon XP 1700+
512 MB DDR-RAM
TerraTec EWX 24/96 soundcard, using alsa-0.9.0_rc2
-- Michael Hellwig aka The Eye olymp.idle.at admin check out http://homepage.uibk.ac.at/~csaa5128 for gpg public key and don't hesitate to look at http://laerm.or.at
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