Transmit Audio + Video - Redistributing Latency

[mudassar]

Hi, 

I am streaming video and audio over rtmp to Wowza. Both video and audio are
captured using appsrc in my application with the following pipeline:

"appsrc name=videosrc format=3 is-live=true do-timestamp=true ! video/x-raw,
width=%ld, height=%ld, framerate=20/1, format=ARGB ! queue
max-size-buffers=1 ! videoconvert  ! vtenc_h264 realtime=true
max-keyframe-interval=60 bitrate=1000 ! h264parse ! flvmux name=mux
streamable=true  ! queue ! rtmpsink name=sink sync=true location=%s  appsrc
name=audiosrc format=3  blocksize=8192 do-timestamp=true is-live=true    !
queue ! audio/x-raw, channels=1, rate=44100, format=F64LE, width=64,
depth=64 ! audioconvert ! audiorate ! audioresample ! audio/x-raw,
channels=1, rate=44100, format=F32LE ! queue ! avenc_aac  ! aacparse ! mux." 

The problem is that audio is always slightly earlier than audio in the live
stream. I have tried adding more queue in audio pipeline, I have tried to
set minimum latency=100000000 at videosrc as well as audiosrc with different
combinations ,nothing worked. 

What should I do ? 




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