bugfix

rtcllm/rtc.py

@@ -5,6 +5,7 @@ from functools import partial
 
 from appPublic.dictObject import DictObject
 from aiortc import MediaStreamTrack, RTCPeerConnection, RTCSessionDescription, RTCIceCandidate
+from aiortc.sdp import candidate_from_sdp, candidate_to_sdp
 from aiortc.contrib.media import MediaBlackhole, MediaPlayer, MediaRecorder, MediaRelay
 
 # from websockets.asyncio.client import connect
@@ -29,6 +30,9 @@ class RTCLLM:
 		self.dc = None
 		self.loop = asyncio.get_event_loop()
 
+	async def ws_send(self, s):
+		self.ws.send(s)
+
 	def get_pc(self, data):
 		return self.peers[data['from'].id].pc
 
@@ -47,15 +51,15 @@ class RTCLLM:
 			self.ws.close()
 
 	async def login(self):
-		await self.ws.send(json.dumps({
+		await self.ws_send(json.dumps({
 			'type':'login',
 			'info':self.info
 		}))
 
 	async def on_icecandidate(self, pc, candidate):
-		print('on_icecandidate()', self, pc, candidate)
+		print('******************** on_icecandidate()', self, pc, candidate)
 		if candidate:
-			await self.ws.send(json.dumps({
+			await self.ws_send(json.dumps({
 				"type":"candidate",
 				"candidate":candidate,
 			}))
@@ -76,14 +80,14 @@ class RTCLLM:
 			'info':data['from'],
 			'pc':pc
 		})
-		await self.ws.send(json.dumps({'type':'callAccepted', 'to':data['from']}))
+		await self.ws_send(json.dumps({'type':'callAccepted', 'to':data['from']}))
 	
 	async def pc_track(self, peerid, track):
 		peer = self.peers[peerid]
 		pc = peer.pc
 		if track.kind == 'audio':
 			f = partial(self.vad_voiceend, peer)
-			vadtrack = AudioTrackVad(track, stage=3, onvoiceend=f)
+			vadtrack = AudioTrackVad(track, stage=0, onvoiceend=f)
 			peer.vadtrack = vadtrack
 			vadtrack.start_vad()
 
@@ -98,6 +102,7 @@ class RTCLLM:
 	async def pc_connectionState_changed(self, peerid):
 		peer = self.peers[peerid]
 		pc = peer.pc
+		print('conn_state={pc.connectionState} ...........')
 		peer.audiotrack = MyAudioTrack()
 		peer.videotrack = MyVideoTrack()
 		pc.addTrack(peer.audiotrack)
@@ -115,8 +120,6 @@ class RTCLLM:
 				k:v for k,v in self.peers.items() if k != peerid
 			}
 			self.peers = peers
-			if len([k for k in self.peers.keys()]) == 0:
-				await self.ws.close()
 
 	def play_video(self, peerid):
 		print('play video ........................')
@@ -146,12 +149,20 @@ class RTCLLM:
 		await pc.setRemoteDescription(offer)
 		answer = await pc.createAnswer()
 		await pc.setLocalDescription(answer)
-		await self.ws.send(json.dumps({
+		await self.ws_send(json.dumps({
 			'type':'answer',
 			'answer':{'type':pc.localDescription.type, 'sdp':pc.localDescription.sdp},
 			'to':data['from']
 		}))
 
+		offer = await pc.createOffer()
+		await pc.setLocalDescription(offer)
+		await self.ws_send(json.dumps({
+			'type':'offer', 
+			'offer': {'type':pc.localDescription.type, 'sdp':pc.localDescription.dsp},
+			'to':data['from']
+		}))
+	
 	async def accept_answer(self, data):
 		pc = self.get_pc(data)
 		answer = RTCSessionDescription(data.answer);
@@ -167,6 +178,7 @@ class RTCLLM:
 		foundation = candidate['candidate'].split(' ')[0]
 		component = candidate['candidate'].split(' ')[1]
 		type = candidate['candidate'].split(' ')[7]
+		"""
 		rtc_candidate = RTCIceCandidate(
 			ip=ip,
 			port=port,
@@ -178,6 +190,10 @@ class RTCLLM:
 			sdpMid=candidate['sdpMid'],
 			sdpMLineIndex=candidate['sdpMLineIndex']
 		)
+		"""
+		rtc_candidate = candidate_from_sdp(candidate['candidate'].split(":", 1)[1])
+		rtc_candidate.sdpMid = candidate['sdpMid']
+		rtc_candidate.sdpMLineIndex = candidate['sdpMLineIndex']
 		await pc.addIceCandidate(rtc_candidate)
 		print('addIceCandidate ok')
 

rtcllm/vad.py

@@ -22,7 +22,7 @@ class AudioTrackVad(MediaStreamTrack):
 		# frameSize = self.track.getSettings().frameSize
 		# self.frame_duration_ms = (1000 * frameSize) / self.sample_rate
 		self.frame_duration_ms = 0.00002
-		self.num_padding_frames = 10
+		self.num_padding_frames = 20
 		self.ring_buffer = collections.deque(maxlen=self.num_padding_frames)
 		self.triggered = False
 		self.voiced_frames = []
@@ -51,12 +51,12 @@ class AudioTrackVad(MediaStreamTrack):
 
 	async def recv(self):
 		oldf = await self.track.recv()
+		self.sample_rate = oldf.sample_rate
 		frames = self.resample(oldf)
 		for f in frames:
 			if self.debug:
 				self.debug = False
 				print(f'{type(f)}, {f.samples=}, {f.format.bytes=}, {f.sample_rate=}, {f.format=}, {f.is_corrupt=}, {f.layout=}, {f.planes=}, {f.side_data=}')
-			self.sample_rate = f.sample_rate
 			try:
 				await self.vad_check(f)
 			except Exception as e:
@@ -69,10 +69,10 @@ class AudioTrackVad(MediaStreamTrack):
 			self.task = self.loop.call_later(self.frame_duration_ms, self._recv)
 		return f
 	
-	def resample(self, frame):
-		fmt = AudioFormat('s16')
-		al = AudioLayout(1)
-		r = AudioResampler(format=fmt, layout=al, rate=frame.rate)
+	def resample(self, frame, sample_rate=None):
+		if sample_rate is None:
+			sample_rate = frame.rate
+		r = AudioResampler(format='s16', layout='mono', rate=sample_rate)
 		frame = r.resample(frame)
 		return frame
 
@@ -104,14 +104,19 @@ class AudioTrackVad(MediaStreamTrack):
 			# audio we've collected.
 			if num_unvoiced > 0.9 * self.ring_buffer.maxlen:
 				self.triggered = False
-				audio_data =  b''.join([self.frame2bytes(f) for f in self.voiced_frames])
+				# audio_data =  b''.join([self.frame2bytes(f) for f in self.voiced_frames])
 				# await self.write_wave(audio_data)
-				await self.gen_base64(audio_data)
+				await self.gen_base64()
 				self.ring_buffer.clear()
 				self.voiced_frames = []
 
 				print('end voice .....', len(self.voiced_frames))
-	async def gen_base64(self, audio_data):
+	async def gen_base64(self):
+		lst = []
+		for f in self.voiced_frames:
+			fs = self.resample(f, sample_rate=16000)
+			lst += fs
+		audio_data =  b''.join([self.frame2bytes(f) for f in lst])
 		b64 = base64.b64encode(audio_data).decode('utf-8')
 		if self.onvoiceend:
 			await self.onvoiceend(b64)