Hi,
I don't know much about the rtp stuff, but shouldn't there
be an 'oggdemux'
after the filesrc?
Stefan
On 1:07:14 pm 18/01/2007 "linguang_wang astrocom.cn" <linguang_wang astrocom.cn> wrote:
> gstreamer-develHi,developers!
> I want to transfer sound file in the real time cross
network. At
> the beginning, I have a test to transfer a ogg file
from
> server to client and then paly it on the client. I
use UDP and RTP .
>
------------------------------------------------------------
-------
> client:
> gst-launch -v filesrc location=qiutianbuhuilai.ogg !
rtpvorbispay !
> udpsink
>
> ERROR: from element /pipeline0/rtpvorbispay0: Element
doesn't
> implement handling of this stream. Please file a bug.
> Additional debug info:
> gstbasertppayload.c(434): gst_basertppayload_push ():
> /pipeline0/rtpvorbispay0: subclass did not specify
clock_rate
> ERROR: pipeline doesn't want to preroll.
>
>
------------------------------------------------------------
-------
> server:
> gst-launch -v udpsrc ! rtpvorbisdepay ! oggdemux !
vorbisdec !
> audioconvert ! osssink
>
> WARNING: erroneous pipeline: could not link
rtpvorbisdepay0 to
> oggdemux0
------------------------------------------------------------
> ---------
> I don't know what's wrong here and what can I do to
play it .
------------------------------------------------------------
-------------
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