On Mon, 2008-07-28 at 13:37 -0700, Dennis Fleming wrote:
> I'm trying to create an audio player on an IMX31 target and I've found
> a discrepancy in the output of various formats.  If I send MP3 data I
> have to set the buffer-time and latency-time to 10000 and 100
> respectively to play without severe dropouts.  However WAV files still
> have drop-out at a consistent rate (about 1 per 10 sec).  Are there
> some general features I'm missing or is there some guidance on the
> buffer-time/latency time that would account for this difference?

What if you just use the defaults? By asking for buffer-time=10000, you
are only providing 10ms buffering in the audio device - if the audio
thread starves for more than 10ms, you are in trouble. Unless you've
taken care to ensure that your kernel provides fine-grained timeslices
and that the system isn't going to be too busy to service the audio
thread, that'll be fine, but unless you really need the low-latency
behaviour (you don't for just playing music), why not set it higher?

J.

> 
> Linux 2.6.22.19
> gstreamer 0.10.17 (open-embedded)
> gst-launch filesrc location=<file> ! decodebin ! alsasink
> buffer-time=10000 latency-time=100
> 
> Dennis
> 
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