Hi,

On Tue, Jul 29, 2008 at 11:15 AM, Zhao Liang-E3423C <[EMAIL PROTECTED]> wrote:
> What's the rootcause of noise and stuttering ?

Now you are using only 1 thread for all the elements and if the
filesrc or the decoder is too slow sometimes
you don't have time to catch up. By adding the queue you put the sink
in another thread and now the filesrc+decoder can
do some decoding in advance.

Gr,
Thijs

>
> For normal playback, it should not have issues. If decoder didn't drop
> data, I think alsasink did it.
> By gstaudiosink mechanism, it will drop data replaced with blank data when
> data is late. I guess the rootcause is that.
>
> If that, I have no ideas except adding a queue before alsasink, and when
> queue is empty, pause the pipeline, it will not cause dropout, but still
> discontinous.
>
> Zhao liang
> ________________________________
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Dennis Fleming
> Sent: Tuesday, July 29, 2008 4:37 AM
> To: gstreamer-embedded@lists.sourceforge.net
> Subject: [gst-embedded] noise and stuttering
>
> 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?
>
> 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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