Subject: Re: [ecasound] buffering question
From: smoerk (smoerk_AT_gmx.de)
Date: Wed Dec 25 2002 - 18:16:19 EET
I'm not the expert in this area, but I hope I can help anyway.
For your purposes there is no need to use ecasound in low latency mode.
The souncards have some buffer to avoid drop outs in the audio, this
means audio output and input is always delayed for some ms. for example:
if you press the play button on your mp3 player you will hear the audio
a little bit later, but it's hardly recognizable (< 200ms).
for some applications it's very important to minimize the delay. if you
use ecasound for real-time effects you need low latency below 10ms. or
for multi-track harddisk recording, if you record the tracks seperatly,
the latency should be as low as possible.
but for your application it's better to use big buffer sizes to avoid
any drop outs.
The Eye wrote:
> 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.
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