From: Itay K. <ik...@gm...> - 2007-04-18 12:03:44
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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. |