[gst-devel] How to use rtpvorbispay and rtpvorbisdepay?

[Wim Taymans]

On Thu, 2007-01-18 at 20:07 +0800, linguang_wang at 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.
You need oggdemux and vorbisparse (for timestamping) like this on thee
server side:

gst-launch -v filesrc location=~/data/JB_FF9_TheGravityOfLove.ogg !
oggdemux ! vorbisparse ! rtpvorbispay ! udpsink

> 
> -------------------------------------------------------------------
> server:
> gst-launch -v udpsrc ! rtpvorbisdepay ! oggdemux !  vorbisdec ! audioconvert ! osssink
> 
On the client side, you need:

gst-launch udpsrc caps="..." ! rtpvorbisdepay ! vorbisdec !
audioconvert ! alsasink sync=0

With caps set to the same value as that on the server side (caps of
udpsink sinkpad). This is usually transmitted using SDP (yes it's huge
for vorbis).

See also:
http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/gst/rtp/README?revision=1.9&view=markup

Wim

> 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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-- 
Wim Taymans <wim at fluendo.com>