WebStudio/server.py

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import asyncio
import websockets
import json
import uuid
import av
import struct
from aiortc import MediaStreamTrack, RTCPeerConnection, RTCSessionDescription
from aiortc.contrib.media import MediaBlackhole, MediaPlayer, MediaRecorder
import jack as Jack
@Jack.set_error_function
def error(msg):
print('Error:', msg)
@Jack.set_info_function
def info(msg):
print('Info:', msg)
jack = Jack.Client('webstudio')
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out1 = jack.outports.register('out_0')
out2 = jack.outports.register('out_1')
def init_buffers():
global transfer_buffer1, transfer_buffer2
transfer_buffer1 = Jack.RingBuffer(jack.samplerate * 10)
transfer_buffer2 = Jack.RingBuffer(jack.samplerate * 10)
init_buffers()
@jack.set_process_callback
def process(frames):
buf1 = out1.get_buffer()
piece1 = transfer_buffer1.read(len(buf1))
buf1[:len(piece1)] = piece1
buf2 = out2.get_buffer()
piece2 = transfer_buffer2.read(len(buf2))
buf2[:len(piece2)] = piece2
class JackSender(object):
def __init__(self, track):
self.track = track
self.resampler = None
async def process(self):
while True:
frame = await self.track.recv()
# Right, depending on the format, we may need to do some fuckery.
# Jack expects all audio to be 32 bit floating point
# while PyAV may give us audio in any format
# (my testing has shown it to be signed 16-bit)
# We use PyAV to resample it into the right format
if self.resampler is None:
self.resampler = av.audio.resampler.AudioResampler(format="fltp", layout="stereo", rate=jack.samplerate)
frame.pts = None # DIRTY HACK
new_frame = self.resampler.resample(frame)
transfer_buffer1.write(new_frame.planes[0])
transfer_buffer2.write(new_frame.planes[1])
current_session = None
class Session(object):
async def end():
await self.pc.close()
init_buffers()
await self.websocket.send(json.dumps({ "kind": "REPLACED" }))
async def connect(self, websocket):
self.websocket = websocket
connection_id = uuid.uuid4();
print(connection_id, "Connected")
await websocket.send(json.dumps({"kind": "HELLO", "connectionId": str(connection_id)}))
sdp_offer = json.loads(await websocket.recv())
if sdp_offer["kind"] != "OFFER":
await websocket.close(1008)
return
offer = RTCSessionDescription(sdp=sdp_offer["sdp"], type=sdp_offer["type"])
print(connection_id, "Received offer")
self.pc = RTCPeerConnection()
self.recorder = MediaRecorder("/home/marks/test.opus")
@self.pc.on("signalingstatechange")
async def on_signalingstatechange():
print(connection_id, "Signaling state is {}".format(self.pc.signalingState))
@self.pc.on("iceconnectionstatechange")
async def on_iceconnectionstatechange():
print(connection_id, "ICE connection state is {}".format(self.pc.iceConnectionState))
if self.pc.iceConnectionState == "failed":
await self.pc.close()
self.pc = None
await websocket.close(1008)
return
@self.pc.on("track")
async def on_track(track):
print(connection_id, "Received track")
if track.kind == "audio":
print(connection_id, "Adding to Jack.")
@track.on("ended")
async def on_ended():
print(connection_id, "Track {} ended".format(track.kind))
init_buffers()
self.sender = JackSender(track)
if current_session is not None:
await current_session.end()
current_session = self
await self.sender.process()
await self.pc.setRemoteDescription(offer)
answer = await self.pc.createAnswer()
await self.pc.setLocalDescription(answer)
await websocket.send(
json.dumps(
{
"kind": "ANSWER",
"type": self.pc.localDescription.type,
"sdp": self.pc.localDescription.sdp,
}
)
)
print(connection_id, "Sent answer")
async for msg in websocket:
print(connection_id, msg)
async def serve(websocket, path):
if path == "/stream":
session = Session()
await session.connect(websocket)
else:
pass
jack.activate()
start_server = websockets.serve(serve, "localhost", 8079)
asyncio.get_event_loop().run_until_complete(start_server)
asyncio.get_event_loop().run_forever()