On Fri, 25 Jan 2002, Matthew Fredrickson wrote:
> Hey all,
>
> I'm working on writing a channel driver for a program called asterisk
> (It's an open source PBX) that the driver will use alsa.
>
> It's been an interesting experience learning alsa, thanks to all who've
> worked on it; It's API is pretty straightforward, though I haven't done
> any sound programming before this.
>
> The circumstance is this: It's a full duplex application so to speak, but
> I don't have control over what order it does reads/writes, etc. I'm only
> basically writing plugin functions for the asterisk channel
> interface. Writes and reads are done in mostly 160 byte (80
> sample, 16bit S-linear) chunks. Also, it's very possible that sometimes
> writes and reads don't come in for extended periods of time (VoIP packeted
> data), so it's not a situation where I know I'm going to get data
> immediately, as opposed to if I was just reading a file or something like
> that.
>
> The first thing I came up with sounded really bad. XRUNs all over the
> place, etc.
>
> I started playing around with various parameters, buffer size, start
> threshold, stuff like that, and it's improved a little bit, but I'm not
> exactly sure why playing around with them has improved it.
>
> The Problem:
> Now I'm at a point where it sounds really good for a few minutes, a
> couple of XRUNS every now and then, and then just about perfect for a
> while. After it has done this, it gets to a point where I rapidly start
> getting more and more XRUNS, and eventually that's about the only thing it
> gets. Boom. No more audio. Arrghhh! :-D
It sound like a problems with bad restarting of a stream. Try to set
start_threshold to buffer_size (it's maximum value). In the case, the
stream is started when the whole ring buffer is filled. It causes a bit
worse latency but a good xrun protection.
Jaroslav
-----
Jaroslav Kysela <[EMAIL PROTECTED]>
SuSE Linux http://www.suse.com
ALSA Project http://www.alsa-project.org
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