Trouble playing a-Law encoded file over RTP/UDP

[Philipp Leibfried]

Hello everyone,

first of all, thanks for the prompt and enlightening answer  to my last question pertaining to GMainLoop and GStreamer.

I'm currently trying to send an a-Law encoded file over RTP and playing it back.
The sender pipeline is setup as follows:

gst-launch-0.10 filesrc location=/home/pl/Projects/gstSender/debug/src/2079.wav do-timestamp=true ! rtppcmapay max-ptime=20000000 ! udpsink host="localhost" port=4044

Before I started using "do-timestamp", sending took only a few seconds, also the file takes about 40 seconds to play back. Now, sending takes about as long as it takes to play back the file, so I guess everything sould be alright on this end(?)

The receiver pipeline is as follows:

gst-launch-0.10 udpsrc port=4044 ! gstrtpjitterbuffer ! rtppcmadepay ! alawdec ! alsasink

After a long long time, I start getting messages such as this:

gstbaseaudiosink.c(1188): gst_base_audio_sink_render (): /pipeline0/alsasink0:
Unexpected discontinuity in audio timestamps of more than half a second (0:02:08.225125000), resyncing
WARNING: from element /pipeline0/alsasink0: Compensating for audio synchronisation problems

And sporadically, I can hear very short bits of the audio (maybe a 10th of a second).

What am I missing here?

Regards and TIA
-Philipp Leibfried

Speech Design GmbH
82110 Germering
Germany
www.speech-design.de




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[Wim Taymans]

On Fri, 2008-08-29 at 15:01 +0200, Philipp Leibfried wrote:
> Hello everyone,
>
> first of all, thanks for the prompt and enlightening answer  to my last question pertaining to GMainLoop and GStreamer.
>
> I'm currently trying to send an a-Law encoded file over RTP and playing it back.
> The sender pipeline is setup as follows:
>
> gst-launch-0.10 filesrc location=/home/pl/Projects/gstSender/debug/src/2079.wav do-timestamp=true ! rtppcmapay max-ptime=20000000 ! udpsink host="localhost" port=4044

do-timestamp on a non-live source does not make sense. Also there is no
way to correctly put a timestamps on random bytes from a file. To get
properly timestamped alaw data you need to put this in a container like
wav and use wavparse to parse and timestamp data.

You could also use something like audioparse on raw int or float audio.

>
> Before I started using "do-timestamp", sending took only a few seconds, also the file takes about 40 seconds to play back. Now, sending takes about as long as it takes to play back the file, so I guess everything sould be alright on this end(?)

That's very likely just coincidence.

>
> The receiver pipeline is as follows:
>
> gst-launch-0.10 udpsrc port=4044 ! gstrtpjitterbuffer ! rtppcmadepay ! alawdec ! alsasink
>
> After a long long time, I start getting messages such as this:
>
> gstbaseaudiosink.c(1188): gst_base_audio_sink_render (): /pipeline0/alsasink0:
> Unexpected discontinuity in audio timestamps of more than half a second (0:02:08.225125000), resyncing
> WARNING: from element /pipeline0/alsasink0: Compensating for audio synchronisation problems

The sender is creating bogus timestamps, the receiver has trouble
handling it.

>
> And sporadically, I can hear very short bits of the audio (maybe a 10th of a second).

coincidence again.

Wim

>
> What am I missing here?
>
> Regards and TIA
> -Philipp Leibfried
>
> Speech Design GmbH
> 82110 Germering
> Germany
> www.speech-design.de
>
>
>
>

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[Philipp Leibfried]

Hi again,

I have now 'repaired' my sender pipeline to look like this (the file I use actually is a WAV container).

gst-launch-0.10 filesrc location=/home/pl/Projects/gstSender/debug/src/2079.wav ! wavparse ! rtppcmapay max-ptime=20000000 ! udpsink host="localhost" port=4044

Wireshark tells me that there is no significant jitter on the RTP "stream" I'm sending (the jitter is 0.01 msec).

However, my receiver pipeline

gst-launch-0.10 udpsrc port=4044 ! rtppcmadepay ! audio/x-alaw, channels=1, rate=8000 ! alawdec ! alsasink

still tells me there is a timestamp disontinuity.

gstbaseaudiosink.c(1188): gst_base_audio_sink_render (): /pipeline0/alsasink0:
Unexpected discontinuity in audio timestamps of more than half a second (0:00:02.049250000), resyncing
WARNING: from element /pipeline0/alsasink0: Compensating for audio synchronisation problems

Forgive my ignorance, but is there something I still need to do on the receiver side? According to Wireshark, my timestamps are good, or is this a misunderstanding on my part?

Thanks
-Philipp


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