Remove outdated file.
authorPhilipp Spitzer <philipp@spitzer.priv.at>
Wed, 21 Apr 2021 18:17:32 +0000 (20:17 +0200)
committerPhilipp Spitzer <philipp@spitzer.priv.at>
Wed, 21 Apr 2021 18:17:32 +0000 (20:17 +0200)
laplace_client.py [deleted file]

diff --git a/laplace_client.py b/laplace_client.py
deleted file mode 100755 (executable)
index 7378c62..0000000
+++ /dev/null
@@ -1,214 +0,0 @@
-#!/usr/bin/python3
-
-import argparse
-import asyncio
-import json
-import logging
-import pathlib
-import ssl
-import sys
-
-import websockets
-
-import gi
-gi.require_version('Gst', '1.0')
-from gi.repository import Gst
-gi.require_version('GstWebRTC', '1.0')
-from gi.repository import GstWebRTC
-gi.require_version('GstSdp', '1.0')
-from gi.repository import GstSdp
-
-
-log = logging.getLogger(__name__)
-
-
-class WebRTCClient:
-
-    def __init__(self, uri: str):
-        self.uri = uri
-        self.ssl_context = ssl.SSLContext()
-        self.ssl_context.check_hostname = False
-        self.ssl_context.verify_mode = ssl.CERT_NONE
-        self.websocket = None
-        self.session_id = None
-        self.userfragments = []
-
-    def send_sdp_offer(self, offer):
-        text = offer.sdp.as_text()
-        log.info(f'send_sdp_offer with {text}')
-        msg = json.dumps({
-            'SessionID': self.session_id,
-            'Type': "gotAnswer",
-            'Value': json.dumps({
-                'type': 'answer',
-                'sdp': text
-            })
-        })
-        loop = asyncio.new_event_loop()
-        loop.run_until_complete(self.websocket.send(msg))
-        loop.close()
-
-    def on_offer_created(self, promise, _, __):
-        log.info('on_offer_created')
-        promise.wait()
-        reply = promise.get_reply()
-        offer = reply.get_value('offer')
-        promise = Gst.Promise.new()
-        self.webrtc.emit('set-local-description', offer, promise)
-        promise.interrupt()
-        self.send_sdp_offer(offer)
-
-        sdp = offer.sdp
-        self.userfragments = [sdp.get_media(i).get_attribute_val('ice-ufrag') for i in range(sdp.medias_len())]
-
-    def on_negotiation_needed(self, element):
-        log.info('on_negotiation_needed')
-        promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
-        element.emit('create-offer', None, promise)
-
-    def send_ice_candidate_message(self, _, mlineindex, candidate):
-        icemsg = json.dumps({
-            'SessionID': self.session_id,
-            'Type': 'addCalleeIceCandidate',
-            'Value': json.dumps({
-                "candidate": candidate,
-                "sdpMid": f"{mlineindex}",
-                "sdpMLineIndex": mlineindex,
-                "usernameFragment": self.userfragments[mlineindex],
-                })
-            })
-        log.info(f'send_ice_candidate_message with {icemsg}')
-        loop = asyncio.new_event_loop()
-        loop.run_until_complete(self.websocket.send(icemsg))
-        loop.close()
-
-    def on_incoming_decodebin_stream(self, _, pad):
-        log.info('on_incoming_decodebin_stream')
-        if not pad.has_current_caps():
-            log.info(pad, 'has no caps, ignoring')
-            return
-
-        caps = pad.get_current_caps()
-        assert (len(caps))
-        s = caps[0]
-        name = s.get_name()
-        if name.startswith('video'):
-            q = Gst.ElementFactory.make('queue')
-            conv = Gst.ElementFactory.make('videoconvert')
-            sink = Gst.ElementFactory.make('autovideosink')
-            self.pipe.add(q, conv, sink)
-            self.pipe.sync_children_states()
-            pad.link(q.get_static_pad('sink'))
-            q.link(conv)
-            conv.link(sink)
-        elif name.startswith('audio'):
-            q = Gst.ElementFactory.make('queue')
-            conv = Gst.ElementFactory.make('audioconvert')
-            resample = Gst.ElementFactory.make('audioresample')
-            sink = Gst.ElementFactory.make('autoaudiosink')
-            self.pipe.add(q, conv, resample, sink)
-            self.pipe.sync_children_states()
-            pad.link(q.get_static_pad('sink'))
-            q.link(conv)
-            conv.link(resample)
-            resample.link(sink)
-
-    def on_incoming_stream(self, _, pad):
-        log.info('on_incoming_stream')
-        if pad.direction != Gst.PadDirection.SRC:
-            return
-        decodebin = Gst.ElementFactory.make('decodebin')
-        decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
-        self.pipe.add(decodebin)
-        decodebin.sync_state_with_parent()
-        self.webrtc.link(decodebin)
-
-    def start_pipeline(self):
-        self.webrtc = Gst.ElementFactory.make('webrtcbin', 'laplace')
-        # self.webrtc.set_property("bundle-policy", 3)
-        direction = GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
-        video_caps = Gst.caps_from_string("application/x-rtp,media=video,encoding-name=VP8/9000,payload=96")
-        audio_caps = Gst.caps_from_string("application/x-rtp,media=audio,encoding-name=OPUS,clock-rate=48000,payload=111")
-        self.webrtc.emit('add-transceiver', direction, video_caps)
-        self.webrtc.emit('add-transceiver', direction, audio_caps)
-        self.pipe = Gst.Pipeline.new("pipeline")
-        Gst.Bin.do_add_element(self.pipe, self.webrtc)
-        self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
-        self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
-        self.webrtc.connect('pad-added', self.on_incoming_stream)
-        self.pipe.set_state(Gst.State.PLAYING)
-        self.webrtc.emit('create-data-channel', 'laplace', None)
-    
-    def close_pipeline(self):
-        self.pipe.set_state(Gst.State.NULL)
-        self.pipe = None
-        self.webrtc = None
-    
-    def handle_sdp(self, sdp):
-        log.info(f'handle_sdp: {sdp}')
-        res, sdpmsg = GstSdp.SDPMessage.new()
-        GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
-        answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
-        promise = Gst.Promise.new()
-        self.webrtc.emit('set-remote-description', answer, promise)
-        promise.interrupt()
-
-    def handle_ice(self, ice):
-        log.info(f'handle_ice: {ice}')
-        candidate = ice['candidate']
-        sdpmlineindex = ice['sdpMLineIndex']
-        self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
-
-    async def run(self):
-        try:
-            async with websockets.connect(self.uri, ssl=self.ssl_context) as websocket:
-                self.websocket = websocket
-                self.start_pipeline()
-                async for msg in websocket:
-                    msg_json = json.loads(msg)
-                    msg_type = msg_json['Type']
-                    msg_value = msg_json['Value']
-                    session_id = msg_json['SessionID']
-                    log.info(f"receive for session {session_id} type {msg_type}")
-                    if msg_type == 'newSession':
-                        self.session_id = session_id
-                    elif msg_type == 'gotOffer':
-                        value_json = json.loads(msg_value)
-                        sdp = value_json['sdp']
-                        self.handle_sdp(sdp)
-                    elif msg_type == 'addCallerIceCandidate':
-                        value_json = json.loads(msg_value)
-                        self.handle_ice(value_json)
-            self.close_pipeline()
-            self.websocket = None
-            self.session_id = None
-        except:
-            log.error(f'Connection to "{self.uri}" failed')
-            sys.exit(1)
-
-
-def check_plugins():
-    for plugin in ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
-                   "rtpmanager", "videotestsrc", "audiotestsrc"]:
-        if Gst.Registry.get().find_plugin(plugin) is None:
-            print('Missing gstreamer plugin:', plugin)
-            return False
-    return True
-
-
-def main():
-    logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
-    Gst.init(None)
-    if not check_plugins():
-        sys.exit(1)
-    parser = argparse.ArgumentParser()
-    parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
-        help='Signalling server URI')
-    args = parser.parse_args()
-    c = WebRTCClient(args.uri)
-    loop = asyncio.get_event_loop()
-    loop.run_until_complete(c.run())
-
-
-if __name__=='__main__':
-    main()