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stream an ogg file over rtp

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Dirk Griffioen-2
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stream an ogg file over rtp

Dirk Griffioen-2
Hi,

I would like to stream an ogg file over rtp

this is my local pipeline, works fine

gst-launch filesrc location=video.ogg ! oggdemux name=d d. ! queue !
theoradec ! ffmpegcolorspace ! xvimagesink d. ! queue ! vorbisdec !
audioconvert ! audioresample ! alsasink

but when I transpose this to rtp it does not work. the pipelines run,
and seem to connect, but the receiving end does not show the video or
sound the audio:


producer:

gst-launch -v gstrtpbin name=rtpbin \
filesrc location=~/Desktop/video.ogg ! oggdemux name=d d. ! queue ! \
    rtptheorapay ! rtpbin.send_rtp_sink_0 \
    rtpbin.send_rtp_src_0 ! udpsink port=10000 host=$REMOTE_HOST \
    rtpbin.send_rtcp_src_0 ! udpsink port=10001 host=$REMOTE_HOST
sync=false async=false    \
    udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0 \
queue ! rtpvorbispay ! rtpbin.send_rtp_sink_1 \
    rtpbin.send_rtp_src_1 ! udpsink port=10003 host=$REMOTE_HOST
ts-offset=0  \
    rtpbin.send_rtcp_src_1 ! udpsink port=10004 host=$REMOTE_HOST
sync=false async=false  \
    udpsrc port=10005 ! rtpbin.recv_rtcp_sink_1


consumer:

gst-launch -v gstrtpbin name=rtpbin latency=20 \
udpsrc caps="application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)THEORA,
sampling=(string)YCbCr-4:2:0, width=(string)640, height=(string)480,
delivery-method=(string)inline, payload=(int)96, \
\
configuration=(string) ...
\
" \
port=10000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! \
rtptheoradepay ! theoradec ! autovideosink \
udpsrc port=10001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! udpsink port=10002 host=$REMOTE_HOST sync=false
async=false \
\
udpsrc
caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)VORBIS,payload=(int)96,\
\
configuration=(string)
\
" \
port=10003 ! rtpbin.recv_rtp_sink_1 rtpbin. ! \
rtpvorbisdepay ! vorbisdec ! jackaudiosink connect=none \
udpsrc port=10004 ! rtpbin.recv_rtcp_sink_1 \
rtpbin.send_rtcp_src_1 ! udpsink port=10005 host=$REMOTE_HOST sync=false
async=false


Does anyone know what I am doing wrong? Not terminating the demuxer 'd.'?

Thanks in advance!

Dirk


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Dirk Griffioen-2
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Re: stream an ogg file over rtp

Dirk Griffioen-2
I managed to do it by reencoding the streams and setting the demux makr
at the correct point

gst-launch -v gstrtpbin name=rtpbin \
filesrc location=~/Desktop/video.ogg ! oggdemux name=d d. !  \
queue ! \
    theoradec ! theoraenc ! rtptheorapay ! rtpbin.send_rtp_sink_0 \
    rtpbin.send_rtp_src_0 ! udpsink port=10000 host=$REMOTE_HOST \
    rtpbin.send_rtcp_src_0 ! udpsink port=10001 host=$REMOTE_HOST
sync=false async=false    \
    udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0 \
\
queue d. ! \
    vorbisdec ! audioresample ! audio/x-raw-float, rate=48000 !
vorbisenc !  rtpvorbispay ! rtpbin.send_rtp_sink_1 \
    rtpbin.send_rtp_src_1 ! udpsink port=10003 host=$REMOTE_HOST
ts-offset=0  \
    rtpbin.send_rtcp_src_1 ! udpsink port=10004 host=$REMOTE_HOST
sync=false async=false  \
    udpsrc port=10005 ! rtpbin.recv_rtcp_sink_1
   
Cheers, Dirk

> Hi,
>
> I would like to stream an ogg file over rtp
>
> this is my local pipeline, works fine
>
> gst-launch filesrc location=video.ogg ! oggdemux name=d d. ! queue !
> theoradec ! ffmpegcolorspace ! xvimagesink d. ! queue ! vorbisdec !
> audioconvert ! audioresample ! alsasink
>
> but when I transpose this to rtp it does not work. the pipelines run,
> and seem to connect, but the receiving end does not show the video or
> sound the audio:
>
>
> producer:
>
> gst-launch -v gstrtpbin name=rtpbin \
> filesrc location=~/Desktop/video.ogg ! oggdemux name=d d. ! queue ! \
>     rtptheorapay ! rtpbin.send_rtp_sink_0 \
>     rtpbin.send_rtp_src_0 ! udpsink port=10000 host=$REMOTE_HOST \
>     rtpbin.send_rtcp_src_0 ! udpsink port=10001 host=$REMOTE_HOST
> sync=false async=false    \
>     udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0 \
> queue ! rtpvorbispay ! rtpbin.send_rtp_sink_1 \
>     rtpbin.send_rtp_src_1 ! udpsink port=10003 host=$REMOTE_HOST
> ts-offset=0  \
>     rtpbin.send_rtcp_src_1 ! udpsink port=10004 host=$REMOTE_HOST
> sync=false async=false  \
>     udpsrc port=10005 ! rtpbin.recv_rtcp_sink_1
>
>
> consumer:
>
> gst-launch -v gstrtpbin name=rtpbin latency=20 \
> udpsrc caps="application/x-rtp, media=(string)video,
> clock-rate=(int)90000, encoding-name=(string)THEORA,
> sampling=(string)YCbCr-4:2:0, width=(string)640, height=(string)480,
> delivery-method=(string)inline, payload=(int)96, \
> \
> configuration=(string) ...
> \
> " \
> port=10000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! \
> rtptheoradepay ! theoradec ! autovideosink \
> udpsrc port=10001 ! rtpbin.recv_rtcp_sink_0 \
> rtpbin.send_rtcp_src_0 ! udpsink port=10002 host=$REMOTE_HOST sync=false
> async=false \
> \
> udpsrc
> caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)VORBIS,payload=(int)96,\
> \
> configuration=(string)
> \
> " \
> port=10003 ! rtpbin.recv_rtp_sink_1 rtpbin. ! \
> rtpvorbisdepay ! vorbisdec ! jackaudiosink connect=none \
> udpsrc port=10004 ! rtpbin.recv_rtcp_sink_1 \
> rtpbin.send_rtcp_src_1 ! udpsink port=10005 host=$REMOTE_HOST sync=false
> async=false
>
>
> Does anyone know what I am doing wrong? Not terminating the demuxer 'd.'?
>
> Thanks in advance!
>
> Dirk
>
>
> ------------------------------------------------------------------------------
> Come build with us! The BlackBerry® Developer Conference in SF, CA
> is the only developer event you need to attend this year. Jumpstart your
> developing skills, take BlackBerry mobile applications to market and stay
> ahead of the curve. Join us from November 9-12, 2009. Register now!
> http://p.sf.net/sfu/devconf
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>  


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is the only developer event you need to attend this year. Jumpstart your
developing skills, take BlackBerry mobile applications to market and stay
ahead of the curve. Join us from November 9-12, 2009. Register now!
http://p.sf.net/sfu/devconf
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