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Decode Redundant Audio Data (RED) as per RFC 2198.
This element is mostly provided for chrome webrtc compatibility: chrome will wrap ulpfec-protected streams in RED packets, and such streams need to be unwrapped by this element before being passed on to GstRtpUlpFecDec.
The “pt” property should be set to the expected payload types of the RED packets.
When using GstRtpBin, this element should be inserted through the “request-aux-receiver” signal.
1 |
gst-launch-1.0 udpsrc port=8888 caps="application/x-rtp, payload=96, clock-rate=90000" ! rtpreddec pt=122 ! rtpstorage size-time=220000000 ! rtpssrcdemux ! application/x-rtp, payload=96, clock-rate=90000, media=video, encoding-name=H264 ! rtpjitterbuffer do-lost=1 latency=200 ! rtpulpfecdec pt=122 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink |
See also: GstRtpRedEnc, GstWebRTCBin, GstRtpBin
plugin |
rtp |
author |
Hani Mustafa <hani@pexip.com>, Mikhail Fludkov <misha@pexip.com> |
class |
Codec/Depayloader/Network/RTP |