Hi,<br><br>I'm trying to do audio streaming using gstreamer, and I can't seem to get it synchronized properly.<br>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?
<br><br>Even in a simple pipeline:<br>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
<br><br>The sound is choppy.<br><br>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:<br><br> gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal, &crate_num, &crate_denom);
<br> clock_offset = (gst_element_get_base_time (GST_ELEMENT_CAST (bsink)) - cexternal) + cinternal;<br><br>suddenly goes from 0 to a very large value, throwing the whole thing out of sync.<br><br>Any hints to what i'm doing wrong? Or how it make it work ok?
<br><br>Thanks,<br>Itay.<br>