Live microphone stream between 2 points

[David Ventura]

> Yes. You need to measure where that latency is introduced. Both alsasrc
> and alsasink will add a non-trivial amount of latency by default. You
> should be able to configure both of these elements for much lower
> latency.


setting latency-time and buffer-time on both ends did nothing for the latency,
reducing buffer-time too much even results in stutter.

I removed both queues (on sender and receiver) which helped a little
but not really.

qos and max-lateness on the receiver didn't help either. I don't know
what else to tweak



On 23 March 2017 at 15:20, David Ventura <davidventura27 at gmail.com> wrote:

> Hi
> I've been trying to listen to a microphone that's on the network but the
> latency is too great for it to be acceptable. We are talking ethernet, ping
> <1ms but actual audio latency is (guessed) ~800ms
>
> My pipeline
>
> gst-launch-1.0 alsasrc device=hw:CARD=CODEC,DEV=0 slave-method=resample provide-clock=true do-timestamp=true buffer-time=20000 ! queue ! audioconvert ! audioresample ! audio/x-raw,format=S16LE,channels=2,rate=48000,layout=interleaved ! multiudpsink clients=192.168.2.208:5003,192.168.2.21:5003
>
>
> And the receiving end
>
> gst-launch-1.0 udpsrc port=5003 ! audio/x-raw,format=S16LE,channels=2,layout=interleaved,rate=$AUDIORATE ! autoaudiosink
>
>
> Is there any way to have very fast LAN audio streaming?
>
>
> David
>
> --
> *Stack* is the new term for "I have no idea what I'm actually using".
>



-- 
*Stack* is the new term for "I have no idea what I'm actually using".
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