My command line code(using gst-launch or shell script) is working fine for Audio and Video over the network.
But ,How to use g_signal_connect() in client side to connect rtpbin with videodepayloader or audiodepayloader.
command line code:
gst-launch -v gstrtpbin name=rtpbin latency=$LATENCY \
udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK \
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=$DEST sync=false async=false
but,In C language when I am writing I am using g_signal_connect() to connect rtpbin with depayloader both case for audio and video.
But,am getting error
new payload on pad: recv_rtp_src_0_511525528_96
**
ERROR:client_H264_AAC.c:92:on_pad_added: assertion failed: (lres == GST_PAD_LINK_OK)
Aborted
Please reply.
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