Streaming uncompressed audio to iOS

[Matt Barclay]

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
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