audio chopped after running for several hours

[Peter Maersk-Moller]

Hi Julien.

Your max-lateness on alsasrc on device 2 *is not* linked to network delay
between device 1 and device 2, but rather linked to an internal clock
started when the first samples are given to alsasrc.

Best regards
Peter


On Thu, Jan 28, 2016 at 8:45 AM, Paixao Julien <J.Paixao at televic.com> wrote:

> Hi,
>
> Would you have any hints regarding my point below?
>
>  > I am now using a network emulator between Device1 and Device2.
> > The IP packets from Device1 to Device2 are first going to the network
> > emulator.
> >
> > With this network emulator I can simulate some network effects.
> > One of them is the jitter and indeed by adding jitter on the line I
> start to have
> > choppy audio since the pipeline of Device2 does not contain a jitter
> buffer
> > before the depayloader.
> >
> > I have also played with the delay parameter, a fixed delay is applied to
> each
> > packet from Device1 to Device2.
> > I have tried a delay of 500ms, 1s and 2s.
> > Let's visualize the situation with a delay of 500ms:
> > Device1 -> Sends RTP packet1 at 0ms   ---------------  Device2 ->
> Receives RTP
> > packet1 after 500ms
> > Device1 -> Sends RTP packet2 at 16ms ---------------  Device2 ->
> Receives RTP
> > packet2 after 516ms
> > Device1 -> Sends RTP packet3 at 16ms ---------------  Device2 ->
> Receives RTP
> > packet3 after 532ms ...
> >
> > In that situation, seeing my Device2 pipeline, I would expect that all
> the
> > packets arriving in Device2 would be dropped by the alsasink since max-
> > lateness is set to 100ms.
> > But the audio on Device2 still plays smoothly, I do not understand that,
> it
> > looks like the timestamp in the RTP packets is not used...
> > Would you have any ideas why?
> >
>
> Best regards,
> Julien.
>
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