rtpmanager: Update codes based on 1.18.4

[platform/upstream/gst-plugins-good.git] / gst / rtpmanager / gstrtprtxqueue.c

diff --git a/gst/rtpmanager/gstrtprtxqueue.c b/gst/rtpmanager/gstrtprtxqueue.c
index a0d6cbe..97678eb 100644 (file)
--- a/gst/rtpmanager/gstrtprtxqueue.c
+++ b/gst/rtpmanager/gstrtprtxqueue.c
@@ -22,10 +22,11 @@
 
 /**
  * SECTION:element-rtprtxqueue
+ * @title: 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
+ * 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:
@@ -45,13 +46,16 @@
  * 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
+ * ]|
+ * 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" ! \
@@ -59,7 +63,8 @@
  *    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
+ * ]|
+ * Receiver pipeline
  */
 
 #ifdef HAVE_CONFIG_H