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webrtc_sendrecv.py: Use sine wave for audio instead of red-noise

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Makes it easier to notice when there's packet loss or other audio
distortion.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742>
This commit is contained in:
Nirbheek Chauhan 2023-01-18 07:32:36 +05:30 committed by GStreamer Marge Bot
parent 492d2b6498
commit 6f99faa080
1 changed files with 1 additions and 1 deletions
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subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
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@ -36,7 +36,7 @@ webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.googl
videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \ videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay picture-id-mode=15-bit ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay picture-id-mode=15-bit !
queue ! application/x-rtp,media=video,encoding-name=VP8,payload={vp8_pt} ! sendrecv. queue ! application/x-rtp,media=video,encoding-name=VP8,payload={vp8_pt} ! sendrecv.
audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={opus_pt} ! sendrecv. queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={opus_pt} ! sendrecv.
''' '''