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rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin
https://bugzilla.gnome.org/show_bug.cgi?id=703426
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2 changed files with 9 additions and 9 deletions
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@ -343,7 +343,7 @@ Some gst-launch-1.0 lines:
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Even when sync is enabled, the two different streams will not play synchronised
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against eachother because the receiver does not have enough information to
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perform this task. For this you need to add the gstrtpbin element in both the
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perform this task. For this you need to add the rtpbin element in both the
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sender and receiver pipeline and use additional sources and sinks to transmit
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RTCP packets used for inter-stream synchronisation.
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@ -18,8 +18,8 @@
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*/
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/**
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* SECTION:element-gstrtpbin
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* @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
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* SECTION:element-rtpbin
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* @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
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*
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* RTP bin combines the functions of #GstRtpSession, #GstRtpSsrcDemux,
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* #GstRtpJitterBuffer and #GstRtpPtDemux in one element. It allows for multiple
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@ -36,7 +36,7 @@
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* the packets are released from the jitterbuffer, they will be forwarded to a
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* #GstRtpPtDemux element. The #GstRtpPtDemux element will demux the packets based
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* on the payload type and will create a unique pad recv_rtp_src_\%u_\%u_\%u on
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* gstrtpbin with the session number, SSRC and payload type respectively as the pad
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* rtpbin with the session number, SSRC and payload type respectively as the pad
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* name.
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*
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* To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_\%u pad. The
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@ -58,7 +58,7 @@
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* mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
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* signal.
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*
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* Access to the internal statistics of gstrtpbin is provided with the
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* Access to the internal statistics of rtpbin is provided with the
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* get-internal-session property. This action signal gives access to the
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* RTPSession object which further provides action signals to retrieve the
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* internal source and other sources.
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@ -67,10 +67,10 @@
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* <title>Example pipelines</title>
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* |[
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* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
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* gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
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* ]| Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
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* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
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* ]| Receive RTP data from port 5000 and send to the session 0 in rtpbin.
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* |[
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* gst-launch-1.0 gstrtpbin name=rtpbin \
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* gst-launch-1.0 rtpbin name=rtpbin \
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* v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
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* rtpbin.send_rtp_src_0 ! udpsink port=5000 \
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* rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
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@ -89,7 +89,7 @@
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* as soon as possible and do not participate in preroll, sync=false and
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* async=false is configured on udpsink
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* |[
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* gst-launch-1.0 -v gstrtpbin name=rtpbin \
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* gst-launch-1.0 -v rtpbin name=rtpbin \
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* udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
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* port=5000 ! rtpbin.recv_rtp_sink_0 \
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* rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
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