How can I send voip data stream alaw encoded with correct data size?

[Uwe Strempel]

Hello,
I've a problem to send the correct data size over voip?
I've following pipeline:
$ gst-launch alsasrc device=hw:1 ! audioconvert ! level name=recordlevel interval=10000000 ! audioconvert ! audioresample ! audio/x-raw-int rate=8000 ! alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=26002

The transmitted rtp -package size is between 512 bytes and 1400 bytes.

How can I force this to 160 bytes or 240 bytes?

Regards
Uwe


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[Wim Taymans]

On 06/14/2011 10:05 AM, Uwe Strempel wrote:

> Hello,
> I've a problem to send the correct data size over voip?
> I've following pipeline:
> $ gst-launch alsasrc device=hw:1 ! audioconvert ! level
> name=recordlevel interval=10000000 ! audioconvert ! audioresample !
> audio/x-raw-int rate=8000 ! alawenc ! rtppcmapay ! udpsink
> host=127.0.0.1 port=26002
>
> The transmitted rtp -package size is between 512 bytes and 1400 bytes.
>
> How can I force this to 160 bytes or 240 bytes?

You can tune the buffer size by setting an appropriate latency-time on
alsasrc.

Wim
>
> Regards
> Uwe
>
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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