[gst-devel] Audio Sinks Clocking Problem?

[Wim Taymans]

On Wed, 2007-04-18 at 15:03 +0300, Itay Kirshenbaum wrote:
> Hi,
> 
> I'm trying to do audio streaming using gstreamer, and I can't seem to
> get it synchronized properly.
> Is there a real problem with using alsasink (or any audio sink for
> that matter) with sync set to true? Or am I just not doing something
> right? 
> 
> Even in a simple pipeline:
> gst-launch audiotestsrc is-live=true ! audioconvert !
> "audio/x-raw-int, width=(int)16, depth=(int)16, signed=(boolean)true,
> endianness=(int)1234, channels=(int)1, rate=(int)8000" ! alsasink
> sync=true 
> 
> The sound is choppy.
> 
> In a more complex pipeline, that does audio RTP streaming, the sound
> starts out alright for the first few samples, and then the
> clock_offset variable:
> 
>     gst_clock_get_calibration (sink->provided_clock, &cinternal,
> &cexternal, &crate_num, &crate_denom); 
>     clock_offset = (gst_element_get_base_time (GST_ELEMENT_CAST
> (bsink)) - cexternal) + cinternal;
> 
> suddenly goes from 0 to a very large value, throwing the whole thing
> out of sync.
> 
> Any hints to what i'm doing wrong? Or how it make it work ok? 

this only started to work reliably in CVS.

Wim

> 
> Thanks,
> Itay.
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