Hi all, I am writing an application for UDP multicast server. I use the below pipelines for server & client. SERVER: gst-launch filesrc location=~/Desktop/h264-aac/alien.mp4 ! qtdemux name=d \ d. ! queue ! rtph264pay ! udpsink port=1234 host=224.0.0.1 \ d. ! queue ! rtpmp4apay ! udpsink port=1233 host=224.0.0.1 CLIENT: gst-launch udpsrc uri=udp://224.0.0.1:1234 caps="application/x-rtp, media=(string)video, encoding-name=(string)H264, payload=(int)96" ! queue ! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! xvimagesink\ udpsrc uri=udp://224.0.0.1:1233 caps="application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)MP4A-LATM, payload=(int)96" ! queue ! rtpmp4adepay ! faad ! audioconvert ! alsasink The pipeline works fine. But when I am writing an application for the above pipeline the pipeline doesn't change its state to "PLAYING". I state change on the pipeline is "GST_STATE_CHANGE_ASYNC". The pipeline never changes its state to "PLAYING". But when if I change my application to send either video or audio packets alone the application works fine with proper state changes. I will be thankful for any help. ------------------------------------------------------------------------------ Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
I could solve the issue. I did not add the queue element in my application program between the qtdemux and payloader element.
Once I added the queue element the server is able to stream properly.
On Wed, Mar 25, 2009 at 2:13 PM, Jyoti D <[hidden email]> wrote:
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