queue problem

[Hoggins!]

Wow ! Thanks a lot ! That did the trick, it's just perfect !

Now I'm facing a strange (well, not so strange) behaviour, maybe there
is something that can be done.

The link providing our RTP stream is not 100% secure. Sometimes, it goes
down. The streaming stops (as expected, of course), but when the link is
back up, the receiving process exits with a "broken pipe" error. I
suspect the shout2send element cannot keep up with the Icecast server
when it is not fed anymore. My only solution is to use an ugly watchdog
script that will restart the GStreamer receiving process whenever it
goes down.

    Hoggins!

Le 02/10/2011 02:16, Mike Mitchell a écrit :
> Hoggins,
>
> If you try "gst-inspect shout2send", you will see that it only has a
> sink.  A queue is just a buffer that creates a thread, but there is
> still only one pipeline and no output from your fitrst shout2send.
> What you need is a 'tee' to divide the pipes into duplicates.  You
> would use queues to thread your encoding. I could not test this
> without your environment, but this should get you close.
>
> gst-launch gstrtpbin name=rtpbin udpsrc port=5000 \
> caps='application/x-rtp, media=(string)audio,
> clock-rate=(int)44100, encoding-name=(string)CELT,
> encoding-params=(string)2, frame-size= (string)480, payload=(int)96'  \
> ! rtpceltdepay ! celtdec ! queue ! audioconvert !  tee name=t \
> t. ! queue ! lame bitrate=320 ! shout2send
> mount=/live320.mp3 port=8000 password=xxxx ip=127.0.0.1
> t. ! queue ! lame bitrate=192 ! shout2send mount=/live320.mp3
> port=8001 password=xxxx ip=127.0.0.1
>
> A diagram of what this might look like is below, it just encodes in
> stereo and mono but is similar.
> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-tee.html
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