Merged yesterday to FFmpeg Git is an initial AV1 RTP packetizer and depacketizer in working to support the tentative AV1 RTP specification.
The AV1 RTP Payload Format is a proposed/draft specification for using the AV1 video codec for purposes from low bit-rate peer-to-peer usage through high bit-rate multi-party video conferencing. This Real-time Transport Protocol (RTP) payload specification is for transmitting video streams using the AV1 video codec and handling both the fragmentation and aggregation of AV1 Open Bitstream Units (OBUs). The draft specification for this AV1 RTP Payload Format can be found via AOMediaCodec’s GitHub.
Merged to FFmpeg Git on Wednesday is an initial AV1 RTP packetizer and depacketizer. It’s sounding off to a great start with the patch from Chris Hodges and is working for AV1 streaming via RTSP between FFmpeg systems and has been tested to work with AV1 RTSP streams with GStreamer:
“Add RTP packetizer and depacketizer according to (most) of the official AV1 RTP specification. This enables streaming via RTSP between ffmpeg and ffmpeg and has also been tested to work with AV1 RTSP streams via GStreamer.
It also adds the required SDP attributes for AV1.
AV1 RTP encoding is marked as experimental due to draft specification status, debug amount reduced and other changes suggested by Tristan.
Added optional code for searching the sequence header to determine the first packet for broken AV1 encoders / parsers.
Stops depacketizing on corruption until next keyframe, no longer prematurely issues packet on decoding if temporal unit was not complete yet.”
Great progress as always in the FFmpeg camp.
