Opus via RTP

[Olivier Crête]

Hi,

What version of GStreamer are you using? You may be using a very old
version. This  gst-launch-1.0 line works fine here on 1.8.x.

Olivier

On Sun, 2016-09-11 at 14:24 +0200, Miha Nedok wrote:
> 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 centricul
ar.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
> > 
> > 
> 
> 
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
-- 
Olivier Crête
olivier.crete at collabora.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20160912/c050f7af/attachment.html>