Bug 218323

Summary:	mediaRecorder dataavailable > websocket > ffmpeg > rtmp
Product:	WebKit	Reporter:	Artus <audrenlaunay>
Component:	WebRTC	Assignee:	Nobody <webkit-unassigned>
Status:	RESOLVED INVALID
Severity:	Normal	CC:	youennf
Priority:	P2
Version:	Safari 14
Hardware:	Unspecified
OS:	Unspecified

Description Artus 2020-10-28 23:04:28 PDT

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.

Comment 1 Alexey Proskuryakov 2020-10-29 17:18:34 PDT

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

Comment 2 youenn fablet 2020-10-30 01:02:31 PDT

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