Hello,
I am capturing an audio input stream with alsasrc and sending it to an iOS app on the LAN over TCP:
gst-launch-1.0 alsasrc ! tcpclientsink host=10.42.0.10 port=5001
And playing it back on iOS:
tcpserversrc port=5001 do-timestamp=true ! audioparse raw-format=4 ! audioconvert ! autoaudiosink
The stream starts out very low latency, around 20-50 ms, but over time the iOS receiver gradually drifts off until the alsasrc sender starts dropping samples.
Is there a way to make the audio sink play back at a slightly faster rate or adjust its playback rate dynamically?
I've tried wrapping in rtpL16pay, but then I also have to use rtpstreampay as this application requires TCP. The processing overhead from those plugins increase latency to an unacceptable range. I am simultaneously capturing video and sending to iOS, and it is always in the 30-60ms range. If the audio stream could reliably stay in the 20-50ms range I could maintain lip sync. Like RTP, encapsulating in a transport stream adds to processing latency, so I'm trying to avoid it.
Thanks,
Matt
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
Locked
