I'm trying to send audio through an RTP Stream using gstreamer with the lowest latency possible and I want to do it from a Pepper(gstreamer 0.10) to my computer(gstreamer 0.10 or 1.0). I can send audio with little latency (20 ms) from the computer to Pepper however I doesn't work as well from the Pepper to the computer. When I try to adjust the buffer-time under 200 ms, I get this type of error :
WARNING: Can't record audio fast enough
Dropped 318 samples. This is most likely beacause downstream can't keep up and is consuming samples too slowly.
I used the answers here and so far and worked with the following pipelines:
Sender
gst-launch-0.10 -v alsasrc name=mic provide-clock=true do-timestamp=true buffer-time=20000 mic. ! \
audio/x-raw-int, format=S16LE, channels=1, width=16,depth=16,rate=16000 ! \
audioconvert ! rtpL16pay ! queue ! udpsink host=pepper.local port=4000 sync=false
Receiver
gst-launch-0.10 -v udpsrc port=4000 caps = 'application/x-rtp, media=audio, clock-rate=16000, encoding-name=L16, encoding-params=1, channels=1, payload=96' ! \
rtpL16depay ! autoaudiosink buffer-time=80000 sync=false
I don't really know how to tackle this issue as the CPU usage is not anormal. And to be frank I am quite new in this, so I don't get what are the parameters to play with to get low latency. I hope someone can help me! (and that it is not a hardware problem too ^^)
Thanks a lot!