[Webkit-unassigned] [Bug 218323] New: mediaRecorder dataavailable > websocket > ffmpeg > rtmp

[bugzilla-daemon at webkit.org]

https://bugs.webkit.org/show_bug.cgi?id=218323

            Bug ID: 218323
           Summary: mediaRecorder dataavailable > websocket > ffmpeg >
                    rtmp
           Product: WebKit
           Version: Safari 14
          Hardware: Unspecified
                OS: Unspecified
            Status: NEW
          Severity: Normal
          Priority: P2
         Component: WebRTC
          Assignee: webkit-unassigned at lists.webkit.org
          Reporter: audrenlaunay at gmail.com
                CC: youennf at gmail.com

Hello,

Is someone have a ffmpeg configuration to receive dataavailable chunks from websocket and transcode it to an rtmp endpoint ?
When using 

Stream #0:1(und): Audio: aac (LC) ([10][0][0][0] / 0x000A), 44100 Hz, mono, fltp, 64 kb/s (default)
Stream #0:0(und): Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuvj420p(pc, progressive), 1280x720, q=-1--1, 0.0000 fps, 1k tbn, 0.0000 tbc (default)

The output need h264 and aac so 

ffmpeg foncfiguration: 

['-i', '-', '-vcodec', 'copy', '-an', rtmpsURL]
- working with iphone
- no sound
- really low output bitrate

['-i', '-', '-c:v', 'libx264', '-preset', 'veryfast', '-tune', 'zerolatency', '-acodec', 'aac', '-ar', '44100', '-b:a', '64k', rtmpsURL]
- no working

Please, if someone has any idea for ffmpeg configuration.

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