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webrtc_sendrecv.py: Add AV1 support when creating the offer
Requires svtav1enc at present for simplicity. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4644>
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commit
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1 changed files with 26 additions and 9 deletions
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@ -34,7 +34,7 @@ except ImportError:
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# These properties all mirror the ones in webrtc-sendrecv.c, see there for explanations
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PIPELINE_DESC_VP8 = '''
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webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302
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videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
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videotestsrc is-live=true pattern=ball ! videoconvert ! queue !
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vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay picture-id-mode=15-bit !
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queue ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt} ! sendrecv.
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audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
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@ -42,14 +42,25 @@ webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302
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'''
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PIPELINE_DESC_H264 = '''
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webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302
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videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
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x264enc tune=zerolatency speed-preset=ultrafast key-int-max=30 intra-refresh=true ! rtph264pay aggregate-mode=zero-latency config-interval=-1 !
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videotestsrc is-live=true pattern=ball ! videoconvert ! queue !
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x264enc tune=zerolatency speed-preset=ultrafast key-int-max=30 intra-refresh=true !
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rtph264pay aggregate-mode=zero-latency config-interval=-1 !
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queue ! application/x-rtp,media=video,encoding-name=H264,payload={video_pt} ! sendrecv.
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audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
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queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
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'''
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# Force I420 because dav1d bundled with Chrome doesn't support 10-bit choma/luma (I420_10LE)
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PIPELINE_DESC_AV1 = '''
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webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302
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videotestsrc is-live=true pattern=ball ! videoconvert ! queue !
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video/x-raw,format=I420 ! svtav1enc preset=13 ! av1parse ! rtpav1pay !
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queue ! application/x-rtp,media=video,encoding-name=AV1,payload={video_pt} ! sendrecv.
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audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
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queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
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'''
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PIPELINE_DESC = {
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'AV1': PIPELINE_DESC_AV1,
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'H264': PIPELINE_DESC_H264,
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'VP8': PIPELINE_DESC_VP8,
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}
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@ -334,22 +345,26 @@ class WebRTCClient:
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self.conn = None
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def check_plugins():
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needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
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def check_plugins(video_encoding):
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needed = ["opus", "nice", "webrtc", "dtls", "srtp", "rtp",
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"rtpmanager", "videotestsrc", "audiotestsrc"]
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if video_encoding == 'vp8':
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needed.append('vpx')
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elif video_encoding == 'h264':
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needed += ['x264', 'videoparsersbad']
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elif video_encoding == 'av1':
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needed += ['svtav1', 'videoparsersbad']
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missing = list(filter(lambda p: Gst.Registry.get().find_plugin(p) is None, needed))
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if len(missing):
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print_error('Missing gstreamer plugins:', missing)
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print_error(f'Missing gstreamer plugins: {missing}')
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return False
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return True
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if __name__ == '__main__':
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Gst.init(None)
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if not check_plugins():
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sys.exit(1)
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parser = argparse.ArgumentParser()
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parser.add_argument('--video-encoding', default='vp8', nargs='?', choices=['vp8', 'h264'],
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parser.add_argument('--video-encoding', default='vp8', nargs='?', choices=['vp8', 'h264', 'av1'],
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help='Video encoding to negotiate')
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parser.add_argument('--peer-id', help='String ID of the peer to connect to')
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parser.add_argument('--our-id', help='String ID that the peer can use to connect to us')
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@ -359,6 +374,8 @@ if __name__ == '__main__':
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dest='remote_is_offerer',
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help='Request that the peer generate the offer and we\'ll answer')
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args = parser.parse_args()
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if not check_plugins(args.video_encoding):
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sys.exit(1)
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if not args.peer_id and not args.our_id:
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print('You must pass either --peer-id or --our-id')
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sys.exit(1)
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