Hello,
I'm trying to play some realtime video sources (web/IP cam) using WebRTC
in a browser. I'm sending RTP stream via Janus gateway using VP9 codec,
hardware transcoded using ffmpeg.
Everything works fine except random frame corruption happening around
moving objects, portions of frame "flowing off the screen" and such
until next keyframe fixes it. This happens consistently especially at
higher framerates.
It turns out the issue could be narrowed down to the VP9 RTP
packetizer. The problem is it's not marking P frames vs I frames in the
VP9 payload descriptor octet (the P bit). Gstreamer does that and
doesn't experience any such corruption issues.
I added this simple change and now WebRTC plays any stream 100% solid
and corruption free for me.
Could somebody implement this simple fix in the upstream. Basically
in libavformat/rtpenc_vp9.c add something to the effect of the following
two lines (to set the P bit for all but I frames):
/* mark the first fragment */
*rtp_ctx->buf_ptr++ = 0x08;
+ if (!keyframe) {
+ rtp_ctx->buf[0] |= 0x40;
Where the "keyframe" is an additional boolean parameter to the
ff_rtp_send_vp9 function which could be called as:
ff_rtp_send_vp9(s1, pkt->data, size, pkt->flags & AV_PKT_FLAG_KEY);
Thanks
Alex
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