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rtpbin2: add a bit of documentation

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2346>
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François Laignel 2025-07-10 19:30:27 +02:00 committed by GStreamer Marge Bot
parent 19e92e5625
commit 618a348b28
2 changed files with 47 additions and 0 deletions
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25
net/rtp/src/rtpbin2/rtprecv.rs
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@ -1,5 +1,30 @@
// SPDX-License-Identifier: MPL-2.0
/**
* SECTION:element-rtprecv
* @see_also: rtpsend, rtpbin, rtpsession, rtpjitterbuffer.
*
* RTP session management (receiver).
*
* ## Example pipeline
*
* |[
* gst-launch-1.0 \
* udpsrc port=5004 caps='application/x-rtp, media=audio, clock-rate=48000, encoding-name=OPUS, encoding-params=(string)1, sprop-stereo=(string)0, payload=96' \
* ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=200000000 ! recv.rtp_sink_0 \
* udpsrc port=5005 caps='application/x-rtcp' \
* ! recv.rtcp_sink_0 \
* rtprecv name=recv rtp-id=example-rtp-id latency=200 \
* ! rtpopusdepay2 ! opusdec ! audioconvert ! audioresample ! queue max-size-bytes=0 max-size-buffers=1 max-size-time=0 ! autoaudiosink \
* rtpsend name=send rtp-id=example-rtp-id \
* send.rtcp_src_0 ! udpsink port=5007 host=127.0.0.1 async=false
* ]| This will process incoming RTP & RTCP packets from UDP ports 5004 & 5005,
* provided the RTP packets contain an Opus encoded audio stream, and will send
* RTCP back to the sender on UDP port 5007.
* See #rtpsend for an example of how to produce such packets.
*
* Since: plugins-rs-0.13.0
*/
use std::collections::{BTreeMap, HashMap};
use std::net::SocketAddr;
use std::ops::ControlFlow;

22
net/rtp/src/rtpbin2/rtpsend.rs
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@ -1,5 +1,27 @@
// SPDX-License-Identifier: MPL-2.0
/**
* SECTION:element-rtpsend
* @see_also: rtprecv, rtpbin, rtpsession, rtpjitterbuffer.
*
* RTP session management (sender).
*
* ## Example pipeline
*
* |[
* gst-launch-1.0 \
* audiotestsrc is-live=true ! opusenc ! rtpopuspay2 ! queue max-size-buffers=0 max-size-bytes=0 max-size-time=100000000 \
* ! rtpsend name=send rtp-id=example-rtp-id \
* send.rtp_src_0 ! udpsink port=5004 host=127.0.0.1 \
* send.rtcp_src_0 ! udpsink port=5005 host=127.0.0.1 async=false \
* rtprecv name=recv rtp-id=example-rtp-id \
* udpsrc port=5007 caps='application/x-rtcp' ! recv.rtcp_sink_0
* ]| This will produce an Opus encoded audio stream and send it as RTP packets with RTCP
* over UDP ports 5004 & 5005. The pipeline expects RTCP to be sent back on UDP port 5007.
* See #rtprecv for an example of how to process such packets.
*
* Since: plugins-rs-0.13.0
*/
use std::collections::HashMap;
use std::pin::Pin;
use std::sync::{Arc, Mutex};