X-Git-Url: https://git.toastfreeware.priv.at/toast/stream2beamer.git/blobdiff_plain/e7f564d0a4e7858c09a49006662c218e133b8462..e539207b2dcd831dd5641bafa75d2a2942f844ce:/lagarde.py diff --git a/lagarde.py b/lagarde.py index 05b2fe0..c33b912 100755 --- a/lagarde.py +++ b/lagarde.py @@ -179,25 +179,57 @@ class WebRTCClient: 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') + 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.props.location = 'rtmp://127.0.0.1:1935/cug' - # sink.props.location = 'rtmp://127.0.0.1:1936/cug' + 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) + + 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) 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) - print(dir(Gst.DebugGraphDetails)) - Gst.debug_bin_to_dot_data(element, Gst.DebugGraphDetails.ALL) + #print(dir(Gst.DebugGraphDetails)) + #Gst.debug_bin_to_dot_data(element, Gst.DebugGraphDetails.ALL) elif name.startswith('audio'): q = Gst.ElementFactory.make('queue') @@ -276,18 +308,6 @@ class WebRTCClient: self.pipe.set_state(Gst.State.NULL) -async def gstreamer_main_loop(): - """Does the equivalent of the following lines in an async friendly way: - loop = GLib.MainLoop() - loop.run() - """ - gst_loop = GLib.MainLoop() - context = gst_loop.get_context() - while True: - events_dispatched = context.iteration(False) - await asyncio.sleep(0. if events_dispatched else 0.01) - - async def run_repeated(task): while True: await task() @@ -297,16 +317,14 @@ async def run_repeated(task): async def run(uri): try: events = Events() - # rtsp = RtspServer() webrtc = WebRTCClient(events) signaling = SignalingClient(events, uri) - main_loop_task = asyncio.Task(gstreamer_main_loop()) - webrtc_task = asyncio.Task(run_repeated(webrtc.run)) - signaling_task = asyncio.Task(run_repeated(signaling.run)) + webrtc_task = asyncio.Task(webrtc.run()) + signaling_task = asyncio.Task(signaling.run()) - done, pending = await asyncio.wait([main_loop_task, webrtc_task, signaling_task], + done, pending = await asyncio.wait([webrtc_task, signaling_task], return_when=asyncio.FIRST_COMPLETED) for task in done: