130 lines
4.2 KiB
Python
130 lines
4.2 KiB
Python
import asyncio
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import websockets
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import json
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import uuid
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import av
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import struct
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from aiortc import MediaStreamTrack, RTCPeerConnection, RTCSessionDescription
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from aiortc.contrib.media import MediaBlackhole, MediaPlayer, MediaRecorder
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import jack as Jack
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@Jack.set_error_function
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def error(msg):
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print('Error:', msg)
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@Jack.set_info_function
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def info(msg):
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print('Info:', msg)
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jack = Jack.Client('webstudio')
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out1 = jack.outports.register('out_1')
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out2 = jack.outports.register('out_2')
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transfer_buffer1 = Jack.RingBuffer(jack.samplerate * 10)
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transfer_buffer2 = Jack.RingBuffer(jack.samplerate * 10)
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@jack.set_process_callback
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def process(frames):
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buf1 = out1.get_buffer()
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piece1 = transfer_buffer1.read(len(buf1))
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buf1[:len(piece1)] = piece1
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buf2 = out2.get_buffer()
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piece2 = transfer_buffer2.read(len(buf2))
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buf2[:len(piece2)] = piece2
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class JackSender(object):
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def __init__(self, track):
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self.track = track
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self.resampler = None
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async def process(self):
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while True:
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frame = await self.track.recv()
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# Right, depending on the format, we may need to do some fuckery.
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# Jack expects all audio to be 32 bit floating point
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# while PyAV may give us audio in any format
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# (my testing has shown it to be signed 16-bit)
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# We use PyAV to resample it into the right format
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if self.resampler is None:
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self.resampler = av.audio.resampler.AudioResampler(format="fltp", layout="stereo", rate=jack.samplerate)
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frame.pts = None # DIRTY HACK
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new_frame = self.resampler.resample(frame)
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transfer_buffer1.write(new_frame.planes[0])
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transfer_buffer2.write(new_frame.planes[1])
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class Session(object):
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async def connect(self, websocket):
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connection_id = uuid.uuid4();
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print(connection_id, "Connected")
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await websocket.send(json.dumps({"kind": "HELLO", "connectionId": str(connection_id)}))
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sdp_offer = json.loads(await websocket.recv())
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if sdp_offer["kind"] != "OFFER":
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await websocket.close(1008)
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return
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offer = RTCSessionDescription(sdp=sdp_offer["sdp"], type=sdp_offer["type"])
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print(connection_id, "Received offer")
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self.pc = RTCPeerConnection()
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self.recorder = MediaRecorder("/home/marks/test.opus")
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@self.pc.on("signalingstatechange")
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async def on_signalingstatechange():
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print(connection_id, "Signaling state is {}".format(self.pc.signalingState))
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@self.pc.on("iceconnectionstatechange")
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async def on_iceconnectionstatechange():
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print(connection_id, "ICE connection state is {}".format(self.pc.iceConnectionState))
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if self.pc.iceConnectionState == "failed":
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await self.pc.close()
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self.pc = None
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await websocket.close(1008)
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return
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@self.pc.on("track")
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async def on_track(track):
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print(connection_id, "Received track")
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if track.kind == "audio":
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print(connection_id, "Adding to Jack.")
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sender = JackSender(track)
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@track.on("ended")
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async def on_ended():
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print(connection_id, "Track {} ended".format(track.kind))
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await sender.process()
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await self.pc.setRemoteDescription(offer)
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answer = await self.pc.createAnswer()
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await self.pc.setLocalDescription(answer)
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await websocket.send(
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json.dumps(
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{
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"kind": "ANSWER",
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"type": self.pc.localDescription.type,
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"sdp": self.pc.localDescription.sdp,
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}
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)
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)
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print(connection_id, "Sent answer")
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async for msg in websocket:
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print(connection_id, msg)
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async def serve(websocket, path):
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if path == "/stream":
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session = Session()
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await session.connect(websocket)
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else:
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pass
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jack.activate()
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start_server = websockets.serve(serve, "localhost", 8079)
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asyncio.get_event_loop().run_until_complete(start_server)
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asyncio.get_event_loop().run_forever()
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