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rtprtxqueue: add basic documentation and example pipelines

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Mostly explaining the difference between rtprtxqueue and rtprtxsend.
This commit is contained in:
George Kiagiadakis 2017-03-20 12:03:29 +02:00
parent 37c491ceff
commit 3e91601fbb
1 changed files with 38 additions and 0 deletions
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38
gst/rtpmanager/gstrtprtxqueue.c
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@ -22,6 +22,44 @@
/**
* SECTION:element-rtprtxqueue
*
* rtprtxqueue maintains a queue of transmitted RTP packets, up to a
* configurable limit (see #GstRTPRtxQueue::max-size-time,
* #GstRTPRtxQueue::max-size-packets), and retransmits them upon request
* from the downstream rtpsession (GstRTPRetransmissionRequest event).
*
* This element is similar to rtprtxsend, but it has differences:
* - Retransmission from rtprtxqueue is not RFC 4588 compliant. The
* retransmitted packets have the same ssrc and payload type as the original
* stream.
* - As a side-effect of the above, rtprtxqueue does not require the use of
* rtprtxreceive on the receiving end. rtpjitterbuffer alone is able to
* reconstruct the stream.
* - Retransmission from rtprtxqueue happens as soon as the next regular flow
* packet is chained, while rtprtxsend retransmits as soon as the retransmission
* event is received, using a helper thread.
* - rtprtxqueue can be used with rtpbin without the need of hooking to its
* #GstRtpBin::request-aux-sender signal, which means it can be used with
* rtpbin using gst-launch.
*
* See also #GstRtpRtxSend, #GstRtpRtxReceive
*
* # Example pipelines
* |[
* gst-launch-1.0 rtpbin name=b rtp-profile=avpf \
* audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! rtprtxqueue ! b.send_rtp_sink_0 \
* b.send_rtp_src_0 ! identity drop-probability=0.01 ! udpsink host="127.0.0.1" port=5000 \
* udpsrc port=5001 ! b.recv_rtcp_sink_0 \
* b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5002 sync=false async=false
* ]| Sender pipeline
* |[
* gst-launch-1.0 rtpbin name=b rtp-profile=avpf do-retransmission=true \
* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \
* b.recv_rtp_sink_0 \
* b. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \
* udpsrc port=5002 ! b.recv_rtcp_sink_0 \
* b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5001 sync=false async=false
* ]| Receiver pipeline
*/
#ifdef HAVE_CONFIG_H