[gst-devel] Lip sync issues in Wowza
martin.secundario at googlemail.com
Tue Feb 23 17:40:17 CET 2010
I've used the following pipeline to send audio (aac) and video (h264) over and RTP/RTCP stream. However, the stream doesn't seems to start.
Could any one give me a hand on it? Btw, how can I response to a concrete thread instead of send new messages everytime?
gst-launch-0.10 -v gstrtpbin name=rtpbin v4l2src device=/dev/video1 ! "video/x-raw-yuv, format=(fourcc)YUY2, width=(int)640, height=(int)480, framerate=(fraction)30/1" ! queue ! videorate ! "video/x-raw-yuv, format=(fourcc)YUY2, width=(int)640, height=(int)480, framerate=(fraction)15/1" ! queue ! ffmpegcolorspace ! "video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)15/1, width=(int)640, height=(int)480" ! queue ! x264enc bitrate=500 ! queue ! rtph264pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink host=192.168.0.230 port=8000 rtpbin.send_rtcp_src_0 ! udpsink host=188.8.131.52 port=8001 sync=false async=false udpsrc port=8001 ! rtpbin.recv_rtcp_sink_0 alsasrc ! "audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2" ! queue ! faac ! queue ! rtpmp4gpay ! rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host=184.108.40.206 port=8002 rtpbin.send_rtcp_src_1 ! udpsink host=220.127.116.11 port=8003 sync=false async=false udpsrc port=8003 ! rtpbin.recv_rtcp_sink_1 udpsink host=192.168.0.230 port=8002 sync=true
Thanks on advance.
> > Hi,
> > I wrote the other day a message to the list but I don't know how to
> > reply again over the same thread.
> > My problem was related to lip sync issues between Gstreamer and the
> > Wowza server.
> > Someone commented that the problem could be on the sdp file and the
> > audio rate. I've checked it and seems to be fine.
> > There are two things that worry me:
> > 1 - The lip sync issues could be related to the fact I'm not using RTCP?
> > 2 - Could be the problem related with the async and sync options used on
> > the pipeline sinks? Btw, could anyone explain me what are these options for?
> > Thank you.
Your right for the (1). If there is no RTCP packet, RTP streams cannot
be synchronized by your client.
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