Bug 218323 – mediaRecorder dataavailable > websocket > ffmpeg > rtmp

• Status: RESOLVED
• Resolution: INVALID
• Component: WebRTC

[Artus]

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.

[Alexey Proskuryakov]

Marking as INVALID as this doesn't appear to be a WebKit bug or enhancement report.

[youenn fablet]

@Artus, which version of Safari are you using?
If you try with no time slice, is your ffmpeg command working?