From: Nirbheek Chauhan Date: Wed, 18 Jan 2023 02:05:18 +0000 (+0530) Subject: webrtc_sendrecv.py: Add support for using H264 encoding X-Git-Tag: 1.22.7~529 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=78c928eefe38d52809620651254afe4661e37264;p=platform%2Fupstream%2Fgstreamer.git webrtc_sendrecv.py: Add support for using H264 encoding Currently only works when we are creating the offer or the offer only contains H264. Part-of: --- diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py index 9878392..79854c5 100755 --- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py +++ b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py @@ -31,14 +31,27 @@ except ImportError: raise # These properties all mirror the ones in webrtc-sendrecv.c, see there for explanations -PIPELINE_DESC = ''' -webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302 +PIPELINE_DESC_VP8 = ''' +webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302 videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \ 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={video_pt} ! sendrecv. 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={audio_pt} ! sendrecv. ''' +PIPELINE_DESC_H264 = ''' +webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302 + videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \ + x264enc tune=zerolatency speed-preset=ultrafast key-int-max=30 intra-refresh=true ! rtph264pay aggregate-mode=zero-latency config-interval=-1 ! + queue ! application/x-rtp,media=video,encoding-name=H264,payload={video_pt} ! sendrecv. + audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! + queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv. +''' + +PIPELINE_DESC = { + 'H264': PIPELINE_DESC_H264, + 'VP8': PIPELINE_DESC_VP8, +} from websockets.version import version as wsv @@ -79,7 +92,7 @@ def get_payload_types(sdpmsg, video_encoding, audio_encoding): class WebRTCClient: - def __init__(self, loop, our_id, peer_id, server, remote_is_offerer): + def __init__(self, loop, our_id, peer_id, server, remote_is_offerer, video_encoding): self.conn = None self.pipe = None self.webrtc = None @@ -93,6 +106,8 @@ class WebRTCClient: self.peer_id = peer_id # Whether we will send the offer or the remote peer will self.remote_is_offerer = remote_is_offerer + # Video encoding: VP8, H264, etc + self.video_encoding = video_encoding.upper() async def send(self, msg): assert self.conn @@ -190,9 +205,9 @@ class WebRTCClient: decodebin.sync_state_with_parent() pad.link(decodebin.get_static_pad('sink')) - def start_pipeline(self, create_offer=True, opus_pt=96, vp8_pt=97): + def start_pipeline(self, create_offer=True, audio_pt=96, video_pt=97): print_status(f'Creating pipeline, create_offer: {create_offer}') - self.pipe = Gst.parse_launch(PIPELINE_DESC.format(vp8_pt=vp8_pt, opus_pt=opus_pt)) + self.pipe = Gst.parse_launch(PIPELINE_DESC[self.video_encoding].format(video_pt=video_pt, audio_pt=audio_pt)) self.webrtc = self.pipe.get_by_name('sendrecv') self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed, create_offer) self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message) @@ -235,10 +250,10 @@ class WebRTCClient: if not self.webrtc: print_status('Incoming call: received an offer, creating pipeline') - pts = get_payload_types(sdpmsg, video_encoding='VP8', audio_encoding='OPUS') - assert('VP8' in pts) + pts = get_payload_types(sdpmsg, video_encoding=self.video_encoding, audio_encoding='OPUS') + assert(self.video_encoding in pts) assert('OPUS' in pts) - self.start_pipeline(create_offer=False, vp8_pt=pts['VP8'], opus_pt=pts['OPUS']) + self.start_pipeline(create_offer=False, video_pt=pts[self.video_encoding], audio_pt=pts['OPUS']) assert(self.webrtc) @@ -316,6 +331,8 @@ if __name__ == '__main__': if not check_plugins(): sys.exit(1) parser = argparse.ArgumentParser() + parser.add_argument('--video-encoding', default='vp8', nargs='?', choices=['vp8', 'h264'], + help='Video encoding to negotiate') parser.add_argument('--peer-id', help='String ID of the peer to connect to') parser.add_argument('--our-id', help='String ID that the peer can use to connect to us') parser.add_argument('--server', default='wss://webrtc.nirbheek.in:8443', @@ -328,7 +345,7 @@ if __name__ == '__main__': print('You must pass either --peer-id or --our-id') sys.exit(1) loop = asyncio.new_event_loop() - c = WebRTCClient(loop, args.our_id, args.peer_id, args.server, args.remote_is_offerer) + c = WebRTCClient(loop, args.our_id, args.peer_id, args.server, args.remote_is_offerer, args.video_encoding) loop.run_until_complete(c.connect()) res = loop.run_until_complete(c.loop()) sys.exit(res)