Streaming audio and video RTP
William Johnston
wgj at cast.uark.edu
Tue Apr 28 18:49:43 UTC 2020
You can only specify ports on element names. Try this:
gst-launch-1.0 -e \
uridecodebin uri="file:///home/fedora/starwars.mov" \
! qtdemux name=demux demux.audio_0 \
! queue \
! audioconvert \
! opusenc bandwidth=superwideband bitrate-type=vbr \
! rtpopuspay \
! rtprtxqueue max-size-time=2000 max-size-packets=0 \
! udpsink host=www.playbacktc.com <http://www.playbacktc.com>
port=5052 \
demux.video_0 \
! queue \
! videorate ! video/x-raw, framerate=30000/1001 \
! videoconvert \
! x264enc tune=zerolatency speed-preset=1 dct8x8=true
quantizer=17 pass=qual \
! rtph264pay \
! rtprtxqueue max-size-time=2000 max-size-packets=0 \
! rtpbin name=rb rb.send_rtp_sink_0 \
! udpsink host=www.playbacktc.com <http://www.playbacktc.com>
port=5054 \
On 4/28/2020 12:42 PM, Patrick Cusack wrote:
> I have a endpoint that expects audio and video over ports 5052 and
> 5054 respectively. I am using the following script to send audio and
> video. I am getting a 'WARNING: erroneous pipeline: syntax error’ when
> I run the command.
> Also, does using simple rtp payloads into a udp sink bypass RTCP
> feedback, ie if my server is NACKing on account of dropped packets,
> does this hinder retransmission of rtp packets?
>
> gst-launch-1.0 -e \
> uridecodebin uri="file:///home/fedora/starwars.mov" \
> ! qtdemux name=demux demux.audio_0 \
> ! queue \
> ! audioconvert \
> ! opusenc bandwidth=superwideband bitrate-type=vbr \
> ! rtpopuspay \
> ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
> ! udpsink host=www.playbacktc.com <http://www.playbacktc.com>
> port=5052 \
> demux.video_0 \
> ! queue \
> ! videorate ! video/x-raw, framerate=30000/1001 \
> ! videoconvert \
> ! x264enc tune=zerolatency speed-preset=1 dct8x8=true
> quantizer=17 pass=qual \
> ! rtph264pay \
> ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
> ! rtpbin.send_rtp_sink_0 \
> ! udpsink host=www.playbacktc.com <http://www.playbacktc.com>
> port=5054 \
>
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