Opus via RTP

[Miha Nedok]

I know that :)

but even if i ad rtpopusdepay i always get the same result.

gst-launch-1.0 -vvvvv udpsrc port=1236
caps="application/x-rtp,media=(string)audio,clock-rate=48000,encoding-params=2,encoding-name=(string)OPUS,payload=111"
 ! rtpopusdepay ! opusdec ! audioconvert
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps =
"application/x-rtp\,\ media\=\(string\)audio\,\ clock-rate\=\(int\)48000\,\
encoding-params\=\(int\)2\,\ encoding-name\=\(string\)OPUS\,\
payload\=\(int\)111"
/GstPipeline:pipeline0/GstRTPOpusDepay:rtpopusdepay0.GstPad:src: caps =
"audio/x-opus\,\ channel-mapping-family\=\(int\)0"
/GstPipeline:pipeline0/GstOpusDec:opusdec0.GstPad:sink: caps =
"audio/x-opus\,\ channel-mapping-family\=\(int\)0"
/GstPipeline:pipeline0/GstRTPOpusDepay:rtpopusdepay0.GstPad:sink: caps =
"application/x-rtp\,\ media\=\(string\)audio\,\ clock-rate\=\(int\)48000\,\
encoding-params\=\(int\)2\,\ encoding-name\=\(string\)OPUS\,\
payload\=\(int\)111"
/GstPipeline:pipeline0/GstOpusDec:opusdec0.GstPad:src: caps =
"audio/x-raw\,\ format\=\(string\)S16LE\,\ layout\=\(string\)interleaved\,\
rate\=\(int\)48000\,\ channels\=\(int\)2\,\
channel-mask\=\(bitmask\)0x0000000000000003"
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps =
"audio/x-raw\,\ format\=\(string\)S16LE\,\ layout\=\(string\)interleaved\,\
rate\=\(int\)48000\,\ channels\=\(int\)2\,\
channel-mask\=\(bitmask\)0x0000000000000003"
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps =
"audio/x-raw\,\ format\=\(string\)S16LE\,\ layout\=\(string\)interleaved\,\
rate\=\(int\)48000\,\ channels\=\(int\)2\,\
channel-mask\=\(bitmask\)0x0000000000000003"
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data
flow error.
Additional debug info:
gstbasesrc.c(2948): gst_base_src_loop ():
/GstPipeline:pipeline0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-linked (-1)
Execution ended after 0:00:00.020207760
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

On Sun, Sep 11, 2016 at 2:11 PM, Sebastian Dröge <sebastian at centricular.com>
wrote:

> On So, 2016-09-11 at 04:17 -0700, MikeSI wrote:
> > I cannot reeive opus via RTP. Even If i set payload it's always the
> > same.
> >
> > gst-launch-1.0 -vvvvv udpsrc port=1236
> > caps="application/x-rtp,media=(string)audio,clock-
> > rate=48000,encoding-params=2,encoding-name=(string)OPUS"
>
> This is not a complete pipeline, you're missing at least the RTP
> depayloader, possibly a decoder and converters, and a sink. E.g.
>
> gst-launch-1.0 -vvvvv udpsrc port=1236 caps="application/x-
> rtp,media=(string)audio,clock-rate=48000,encoding-params=2,encoding-
> name=(string)OPUS" ! rtpopusdepay ! opusdec ! fakesink
>
> --
> Sebastian Dröge, Centricular Ltd · http://www.centricular.com
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20160911/f83ec4a2/attachment-0001.html>