GStreamer: Add dummy audio track to the received rtp stream

﹥>﹥吖頭↗ 提交于 2019-12-24 00:51:58

问题


I'm initiating RTP stream from my Raspberry camera using:

raspivid -n -vf -fl -t 0 -w 640 -h 480 -b 1200000 -fps 20 -pf baseline -o - | gst-launch-1.0 -v fdsrc ! h264parse ! rtph264pay pt=96 config-interval=10 ! udpsink host=192.168.2.3 port=5000

on the client site, I'm converting it to HLS and upload it on a web server:

gst-launch-1.0 udpsrc port=5000 ! application/x-rtp,payload=96 ! rtph264depay ! mpegtsmux ! hlssink max-files=5 target-duration=5 location=C:/xampp/htdocs/live/segment%%05d.ts playlist-location=C:/xampp/htdocs/live/playlist.m3u8

the above works with me. on the other hand some players does not play the HLS since it has no audio track. I'm trying to figure out how i can add a dummy audio track. I tried many things but no luck e.g.

gst-launch-1.0 udpsrc port=5000 ! application/x-rtp,payload=96 ! rtph264depay ! h264parse ! mux. audiotestsrc wave=4 freq=200 ! audioconvert ! queue ! mux. mpegtsmux name=mux ! hlssink max-files=5 target-duration=5 location=C:/xampp/htdocs/live/segment%%05d.ts playlist-location=C:/xampp/htdocs/live/playlist.m3u8

or

gst-launch-1.0 -e -v udpsrc port=5000 name=src ! application/x-rtp,payload=96 ! rtph264depay ! h264parse ! mpegtsmux name=mux ! audiotestsrc wave=silence src. ! audioconvert ! wavenc ! rtpmp4gdepay ! aacparse ! mux. ! hlssink max-files=5 target-duration=5 location=C:/xampp/htdocs/live/segment%%05d.ts playlist-location=C:/xampp/htdocs/live/playlist.m3u8

Any help is appreciated


回答1:


What was your idea for these pipelines? These look like you are trying to mux uncompressed audio data. I don't think this is what you want. I expected something like this for the audio path:

audiotestsrc wave=silence ! voaacenc ! aacparse ! mux.

Note that there may be more specific requirements - like number of audio channels or specific sample rates that are supported by your HLS players.



来源:https://stackoverflow.com/questions/37904594/gstreamer-add-dummy-audio-track-to-the-received-rtp-stream

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