X-Git-Url: https://git.toastfreeware.priv.at/toast/stream2beamer.git/blobdiff_plain/3a4448810c5ee5360befc23b8e27fa04bcc86015..0831639af5c4d7954b3727cabcb7a75e0bd0bdf7:/lagarde.py diff --git a/lagarde.py b/lagarde.py index e9fca23..fd82852 100755 --- a/lagarde.py +++ b/lagarde.py @@ -177,24 +177,74 @@ class WebRTCClient: if not pad.has_current_caps(): log.info(pad, 'has no caps, ignoring') return - caps = pad.get_current_caps() padsize = caps.get_size() + + log.info(f'>>>> {padsize} {caps}') + for i in range(padsize): s = caps.get_structure(i) # Gst.Structure name = s.get_name() + log.info(f'###### {name}') if name.startswith('video'): q = Gst.ElementFactory.make('queue') conv = Gst.ElementFactory.make('videoconvert') - sink = Gst.ElementFactory.make('intervideosink') + enc = Gst.ElementFactory.make('x264enc') + enc.set_property('bitrate', 1000) + enc.set_property('tune', 'zerolatency') + capsfilter = Gst.ElementFactory.make('capsfilter') + capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc')) + flmux = Gst.ElementFactory.make('flvmux') + sink = Gst.ElementFactory.make('rtmpsink') + sink.set_property('location', 'rtmp://192.168.1.46:1935/gregoa') + # sink.set_property('location', 'rtmp://bla:1936/gregoa') + print(sink.props.location, dir(sink.props)) + assert q and conv and enc and capsfilter and flmux and sink + self.pipe.add(q) self.pipe.add(conv) + self.pipe.add(enc) + self.pipe.add(capsfilter) + self.pipe.add(flmux) + self.pipe.add(sink) + self.pipe.sync_children_states() + + q_pad_sink = q.get_static_pad('sink') + assert q_pad_sink + pad_link_return = pad.link(q_pad_sink) + assert pad_link_return == Gst.PadLinkReturn.OK + + # ok = element.link(q) + # assert ok + + ok = q.link(conv) + assert ok + ok = conv.link(enc) + assert ok + ok = enc.link(capsfilter) + assert ok + ok = capsfilter.link(flmux) + assert ok + ok = flmux.link(sink) + assert ok + self.pipe.set_state(Gst.State.PLAYING) + #print(dir(Gst.DebugGraphDetails)) + #Gst.debug_bin_to_dot_data(element, Gst.DebugGraphDetails.ALL) + + 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) + self.pipe.add(conv) + self.pipe.add(resample) self.pipe.add(sink) self.pipe.sync_children_states() pad.link(q.get_static_pad('sink')) q.link(conv) - conv.link(sink) - self.pipe.set_state(Gst.State.PLAYING) + conv.link(resample) + resample.link(sink) def set_remote_desciption_done(self, gst_promise): gst_promise = Gst.Promise.new_with_change_func(self.create_answer_done) @@ -280,7 +330,7 @@ async def run(uri): try: events = Events() - rtsp = RtspServer() + # rtsp = RtspServer() webrtc = WebRTCClient(events) signaling = SignalingClient(events, uri) @@ -300,7 +350,7 @@ async def run(uri): def main(): - logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s') + logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s') parser = argparse.ArgumentParser() parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug', help='Signalling server URI')