X-Git-Url: https://git.toastfreeware.priv.at/toast/stream2beamer.git/blobdiff_plain/d4e4bdcc8b04d12fc1e38922230b70fb2e047d82..e97741140341d1c98f6cb350d819a988c8b73bbd:/lagarde.py

diff --git a/lagarde.py b/lagarde.py
index 94ea57e..f31bc2d 100755
--- a/lagarde.py
+++ b/lagarde.py
@@ -170,16 +170,17 @@ class WebRTCClient:
                 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')
+                flvmux = Gst.ElementFactory.make('flvmux')
+                flvmux.set_property('streamable', True)
                 sink = Gst.ElementFactory.make('rtmpsink')
                 sink.set_property('location', self.rtmp_uri)
-                assert q and conv and enc and capsfilter and flmux and sink
+                assert q and conv and enc and capsfilter and flvmux 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(flvmux)
                 self.pipe.add(sink)
 
                 q_pad_sink = q.get_static_pad('sink')
@@ -193,9 +194,9 @@ class WebRTCClient:
                 assert ok
                 ok = enc.link(capsfilter)
                 assert ok
-                ok = capsfilter.link(flmux)
+                ok = capsfilter.link(flvmux)
                 assert ok
-                ok = flmux.link(sink)
+                ok = flvmux.link(sink)
                 assert ok
                 self.pipe.set_state(Gst.State.PLAYING)
                 self.pipe.sync_children_states()
@@ -222,22 +223,28 @@ class WebRTCClient:
     def create_answer_done(self, gst_promise):
         reply = gst_promise.get_reply()
         answer = reply.get_value('answer')
+        gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done)
+        self.webrtcbin.emit('set-local-description', answer, gst_promise)
+
         sdp_message = answer.sdp
         mids = [sdp_message.get_media(i).get_attribute_val('mid')
                 for i in range(sdp_message.medias_len())]
         user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
                           for i in range(sdp_message.medias_len())]
-        self.events.sdp_info.put_nowait((mids, user_fragments))
         sdp_answer = sdp_message.as_text()
-        log.info(f'Send SDP answer')
-        log.debug(f'SDP answer:\n{sdp_answer}')
-        self.events.sdp_answer.put_nowait(sdp_answer)
-        gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done)
-        self.webrtcbin.emit('set-local-description', answer, gst_promise)
+        self.mids_uf = mids, user_fragments
+        self.answer = sdp_answer
 
     def set_local_description_done(self, gst_promise):
         gst_promise.get_reply()
 
+        sdp_answer = self.answer
+        log.info(f'Send SDP answer')
+        log.debug(f'SDP answer:\n{sdp_answer}')
+        self.events.sdp_answer.put_nowait(sdp_answer)
+        mids, user_fragments = self.mids_uf
+        self.events.sdp_info.put_nowait((mids, user_fragments))
+
     async def run(self):
         bus = Gst.Pipeline.get_bus(self.pipe)
         self.pipe.set_state(Gst.State.PLAYING)
@@ -253,11 +260,8 @@ class WebRTCClient:
                         return
                 elif self.events.sdp_offer.qsize() > 0:
                     sdp_offer = self.events.sdp_offer.get_nowait()
-                    res, sm = GstSdp.SDPMessage.new()
+                    res, sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
                     assert res == GstSdp.SDPResult.OK
-                    GstSdp.sdp_message_parse_buffer(bytes(sdp_offer.encode()), sm)
-                    # the three lines above can also be done this way in new versions of GStreamer:
-                    # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
                     rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
                     gst_promise = Gst.Promise.new_with_change_func(self.set_remote_desciption_done)
                     self.webrtcbin.emit('set-remote-description', rd, gst_promise)