a poor man's road to multitrack recording ...

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Subject: a poor man's road to multitrack recording ...
From: Carsten Bauer (cbauer-@t-online.de)
Date: Thu Mar 20 2003 - 18:33:36 EET

Hi all,

I've recently started to explore sound processing. Got me an Creative SB
Audigy card, updated to suse Linux 8.1, and thereby noticed that I have an
onbord sound card, too (on my epox 8kt3a board)., didn't notice it before..

In contrast to the ALSA tool arecord, ecasound enabled me to get recordings
without buffer overruns (-z:db is very helpful !), so I started the next
step: multitrack recording. The version form the examples-docs (modified for

-a:1 -i monitor.wav -o alsa,emu10k1
-a:2 -i alsa,emu10k1 -o new-file.wav

works perfect, but I wanted to get the second track separately, not mixed
together with the monitor. Inspired by some recent emails in this list I
tried to open a 4 channel file

-a:1 -i monitor.wav -o alsa,emu10k1
-a:2 -f:16,4,44100 -i alsa,emu10k1 -erc:1,3 -erc:2,4 -o new4channelfile.wav

but alsa-lib complains about the 4 channel setting (channel count out of
I only managed to play back the monitor in mono on the right channel and
record separately in mono on the left channel .

now, finally, my questions:
Is there any other way to get stereo multitrack recording with the same card
and the new track separated from the monitor ?

A general issue: when recording, the signal is always automatically forwarded
to the output, e.g, if a mic is plugged in, you already hear its sound from
the speakers even if no recording process is started. Is there any way to
block this? (Just turning down the mixer volume is of no use, as then the
monitor track cannot be heared anymore..)

I tried multitrack recording with my two cards,

-a:1 -i monitor.wav -o alsa,via686
-a:2 -i alsa,emu10k1 -o new-file.wav

playing something on the via686 and recording with the audigy card, this
works, giving me separate files. But later, when putting them together, I
have to cut away the first, say, 0.12 seconds from the recorded file to
account for the latency of the playback
now I would like to know how accurate ecasound woks internally, e.g when
specifying -gc:0.115,240 ecasound responds:
Setting parameters: open-at-sec = 0.12, duration-sec = 240.00
Is this truncation only for the display, or performed for internal calc. too?
Maybe, accuracy to 0.01s is adequate for this type of equipment and shorter
values would make no sense anyway. Hmm?

Any tips from the sound gurus are welcome !!

So long

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