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webrtc-sendrecv.py: improve debug and documentation
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2 changed files with 8 additions and 2 deletions
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@ -74,6 +74,12 @@ $ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstrea
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* python3 -m pip install --user websockets
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* run `python3 sendrecv/gst/webrtc-sendrecv.py ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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> The python version currently requires the master branches from `gst-plugins-bad` and `gst-plugins-base`.
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<!---
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TODO: replace the note above when 1.16 is released
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-->
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With all versions, you will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app.
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You can pass a --server argument to all versions, for example `--server=wss://127.0.0.1:8443`.
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@ -42,7 +42,7 @@ class WebRTCClient:
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def send_sdp_offer(self, offer):
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text = offer.sdp.as_text()
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print ('Sending offer:\n%s', text)
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print ('Sending offer:\n%s' % text)
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msg = json.dumps({'sdp': {'type': 'offer', 'sdp': text}})
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loop = asyncio.new_event_loop()
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loop.run_until_complete(self.conn.send(msg))
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@ -120,7 +120,7 @@ class WebRTCClient:
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sdp = msg['sdp']
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assert(sdp['type'] == 'answer')
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sdp = sdp['sdp']
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print ('Received answer:\n%s', sdp)
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print ('Received answer:\n%s' % sdp)
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res, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp)
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answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
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promise = Gst.Promise.new()
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