Add audio and record RTP stream

[Andrew Borntrager]

Ok, so this sends audio: (wireshark goes nuts on windows laptop, port 5002
and raspberry pi runs gstreamer with no errors)
gst-launch-1.0 -v pulsesrc
device="alsa_input.usb-046d_webcam.........stereo" ! opusenc ! rtpopuspay
pt=96 ! udp sink host=xxxxxx.net port 5002.

This gives me "WARNING Erroneous pipleline: could not link udpsink0 to
pulsesrc0" :
gst-launch-1.0 -v vl2src device=/dev/video0 ! video/x-264,
width=1280,height=720, framerate=15/1 ! h264parse ! rtph264pay pt=127
config-interval=4 ! udpsink host=xxxxxx,net port=5000! pulsesrc
device="alsa_input.usb-046d_webcam.........stereo" ! opusenc ! rtpopuspay
pt=96 ! udp sink host=xxxxxx.net port 5002.

But this works: (for video)
gst-launch-1.0 -v vl2src device=/dev/video0 ! video/x-264,
width=1280,height=720, framerate=15/1 ! h264parse ! rtph264pay pt=127
config-interval=4 ! udpsink host=xxxxxx,net port=5000

This doesn't work for receiving audio (even though wireshark says im
getting something on 5002:
gst-launch-1.0 udpsrc
caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)X-GST-OPUS-DRAFT-SPITTKA-00,
payload=(int)96,ssrc=(uint)559994649" port=5002 ! rtpbin ! rtpopusdepay !
opusdec !audioconvert ! audioresample ! autoaudiosink

Tried autoaudiosink and directsoundsink, run with no errors, but just sits
there. Thanks for any help!



On Tue, Jul 12, 2016 at 2:29 AM, Sebastian Dröge <sebastian at centricular.com>
wrote:

> On Fr, 2016-07-08 at 10:53 -0400, Andrew Borntrager wrote:
> > Hello again.
> > I *think* I'm sending audio along with video. The pipeline is running
> > (I changed to ports 5000 and 5002) with no errors.  Now I'm trying
> > this on my windows laptop.(doesn't work):
> >
> > gst-launch-1.0  udpsrc caps="application/x-rtp, media=(string)video,
> > clock-rate=(int) 90000, encoding-name=(string)H264,
> > sampling=(string)YCbCr-4:4:4, depth=(string)8, width=(string)320,
> > height=(string)240, payload=(int)96, clock-base=(uint)4068866987,
> > seqnum-base=(uint)24582" port=5000 ! rtph264depay ! decodebin !queue!
> > autovideosink ! udpsrc caps=$AUDIO_CAPS port=5002 !
> > gstrtpjitterbuffer! rtpopusdepay ! opusdec plc=true ! alsasink
>
> There is no alsasink on Windows, try using directsoundsink or just
> autoaudiosink if you want to use the same code on all platforms. Also
> you should insert videoconvert ! videoscale before autovideosink, and
> audioconvert ! audioresample before the audio sink.
>
> If that doesn't help, how does it not work? You might also want to set
> a bigger value on the buffer-size property on the udpsrcs, especially
> for raw video.
>
> --
>
> 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
>
>
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