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Thread: Audio Sinks Clocking Problem?
Audio Sinks Clocking Problem?
user name
 2007-04-18 07:03:39 

  


  

    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?


Thanks,
Itay.



  

  
  
   
     
    






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