[ecasound] Ecaplayer-RTprio-Volumecontrol-Buffer

From: Klaus Schulz <klaus.schulz@email-addr-hidden>
Date: Mon Feb 12 2007 - 15:27:07 EET

Hi there.

I am new to the list. Though I am using ecasound for a couple of weeks now. And I am very happy with it.

I am trying to set-up ecasound as a state of-the-art audio-engine for high-end playback first.
Soundwise I got very far already. I havn't listened to any audio-engine playing better then ecasound
the way I use it. So far so good.

Since Ecaplay doesn't offer all the options to do it, I am using ecasound instead.
The interactive mode is not usable since the options I need are not supported.
I am talking especially about DSP options. E.g. volume control using the -ea function.
And an intelligent playlist handling is also missing.

I've written a script, which let me easily find tracks by using the find command and
I can control the volume with -ea. (Not interactive, but between the change of tracks!)

I am starting ecasound like this (framed by a script!):

ecasound -r:80 -b:2 -B:rtlowlatency -i:alsahw,1,0 -ea:50 -o:alsahw,1,0

Questions to understand the whole subject a bit better:

1.a The rt-priority is not shown when using "ps -C ecasound -o rtprio". How do I know if I run realtime at what parameter?
     Other programs using rtpriority clearly show this with above command!
     (I am running it as root of course.)
1.b Are actually Alsa, the soundcard driver and other process that might be in the chain working in realtime?
       How do I make sure that this is the case?
2. With buffersize of two samples I can run ecasound without xruns on my PC. I am wondering if I can trust this buffer setting I am using.
    Alsa seems to complain that the buffersize doesn't match the period size ( I read that the period size is supposed to be 1/4th
    of the buffer size) I could not find a paramter to set the period size of alsa within ecasound.
    This low buffer size is giving me best sound results!
3.a. Volumecontrol: A scale in db would be helpful, and it should be called attentuation insted of amplification!
    What is strange though: I need to get the parameter as low as 15 to 20% for running acceptable volume levels in my setup, running at sound levels of 95db or so.
   Any explanations would be helpful.
3.b. With e.g. Samplitude I am running -12--20db to have a similar level. What algorithm is behind ecasound scale?
3.c. I assume that you do 32bit float volume control with the DSP. Will the result be downsized to 16bit before sent to te output?
       Is there a chance to dither the signal?
4. Buffering of tracks: I am buffering all tracks on /tmp (tmpfs) in RAM before playing back, that improves the sound heavily.
    Is there an option to do full track buffering with ecasound?

Where would I have to start to build myself an extended "ecaplayer" using existing valubale APIs. (I am not very experienced in programming yet!)

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Received on Mon Feb 12 16:15:01 2007

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