Add audio and record RTP stream

[Andrew Borntrager]

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

Also, i would like to simultaneously record the stream on my laptop. Thank you very much for the overwhelming support! 

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[Sebastian Dröge-3]

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
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[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 <[hidden email]> 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
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[Nicolas Dufresne-4]

Le mercredi 13 juillet 2016 à 20:31 -0400, Andrew Borntrager a écrit :
> 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.

There is a ! between udpsink and pulsesrc, remove it. You don't want
any link between audio and video, you just want them in the same
pipeline for sync.

regards,
Nicolas
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