Closed ShevKG closed 2 years ago
Logs looks like this:
Blob { size: 65580, type: "audio/wav" }
ggs:269:29
Blob { size: 57388, type: "audio/wav" }
ggs:269:29
Blob { size: 57388, type: "audio/wav" }
ggs:269:29
Blob { size: 57388, type: "audio/wav" }
ggs:269:29
Blob { size: 65580, type: "audio/wav" }
ggs:269:29
Blob { size: 204844, type: "audio/wav" } <--- this blob is much bigger than other's (timeslice is on 500ms)
ggs:269:29
Blob { size: 57388, type: "audio/wav" }
On the next day a problem has gone. I think i should have just refresh the browser and do not testing p2p audio stream simultaneously on a single computer.
I'm struggling working around with
timeSlice()
method of RecordRTC. It looks like sometimes it freeze for a second and after thatshows in logs that recorded blob is much bigger than others. I tried to reset and release buffered memory, but it didn't worked. Can i even track why this is happening somehow?