Files
gradio-webrtc/backend/fastrtc/websocket.py
huangbinchao.hbc aefb08150f [feat] update some feature
sync code of  fastrtc,
add text support through datachannel,
fix safari connect problem
support chat without camera or mic
2025-03-25 18:05:10 +08:00

216 lines
7.9 KiB
Python

import asyncio
import audioop
import base64
import logging
from typing import Any, Awaitable, Callable, Optional, cast
import anyio
import librosa
import numpy as np
from fastapi import WebSocket
from .tracks import AsyncStreamHandler, StreamHandlerImpl
from .utils import AdditionalOutputs, DataChannel, split_output
class WebSocketDataChannel(DataChannel):
def __init__(self, websocket: WebSocket, loop: asyncio.AbstractEventLoop):
self.websocket = websocket
self.loop = loop
def send(self, message: str) -> None:
asyncio.run_coroutine_threadsafe(self.websocket.send_text(message), self.loop)
logger = logging.getLogger(__file__)
def convert_to_mulaw(
audio_data: np.ndarray, original_rate: int, target_rate: int
) -> bytes:
"""Convert audio data to 8kHz mu-law format"""
if audio_data.dtype != np.float32:
audio_data = audio_data.astype(np.float32) / 32768.0
if original_rate != target_rate:
audio_data = librosa.resample(audio_data, orig_sr=original_rate, target_sr=8000)
audio_data = (audio_data * 32768).astype(np.int16)
return audioop.lin2ulaw(audio_data, 2) # type: ignore
run_sync = anyio.to_thread.run_sync # type: ignore
class WebSocketHandler:
def __init__(
self,
stream_handler: StreamHandlerImpl,
set_handler: Callable[[str, "WebSocketHandler"], Awaitable[None]],
clean_up: Callable[[str], None],
additional_outputs_factory: Callable[
[str], Callable[[AdditionalOutputs], None]
],
):
self.stream_handler = stream_handler
self.stream_handler._clear_queue = self._clear_queue
self.websocket: Optional[WebSocket] = None
self._emit_task: Optional[asyncio.Task] = None
self.stream_id: Optional[str] = None
self.set_additional_outputs_factory = additional_outputs_factory
self.set_additional_outputs: Callable[[AdditionalOutputs], None]
self.set_handler = set_handler
self.quit = asyncio.Event()
self.clean_up = clean_up
self.queue = asyncio.Queue()
def _clear_queue(self):
old_queue = self.queue
self.queue = asyncio.Queue()
logger.debug("clearing queue")
i = 0
while not old_queue.empty():
try:
old_queue.get_nowait()
i += 1
except asyncio.QueueEmpty:
break
logger.debug("popped %d items from queue", i)
def set_args(self, args: list[Any]):
self.stream_handler.set_args(args)
async def handle_websocket(self, websocket: WebSocket):
await websocket.accept()
loop = asyncio.get_running_loop()
self.loop = loop
self.websocket = websocket
self.data_channel = WebSocketDataChannel(websocket, loop)
self.stream_handler._loop = loop
self.stream_handler.set_channel(self.data_channel)
self._emit_task = asyncio.create_task(self._emit_loop())
self._emit_to_queue_task = asyncio.create_task(self._emit_to_queue())
if isinstance(self.stream_handler, AsyncStreamHandler):
start_up = self.stream_handler.start_up()
else:
start_up = anyio.to_thread.run_sync(self.stream_handler.start_up) # type: ignore
self.start_up_task = asyncio.create_task(start_up)
try:
while not self.quit.is_set():
message = await websocket.receive_json()
if message["event"] == "media":
audio_payload = base64.b64decode(message["media"]["payload"])
audio_array = np.frombuffer(
audioop.ulaw2lin(audio_payload, 2), dtype=np.int16
)
if self.stream_handler.input_sample_rate != 8000:
audio_array = audio_array.astype(np.float32) / 32768.0
audio_array = librosa.resample(
audio_array,
orig_sr=8000,
target_sr=self.stream_handler.input_sample_rate,
)
audio_array = (audio_array * 32768).astype(np.int16)
if isinstance(self.stream_handler, AsyncStreamHandler):
await self.stream_handler.receive(
(self.stream_handler.input_sample_rate, audio_array)
)
else:
await run_sync(
self.stream_handler.receive,
(self.stream_handler.input_sample_rate, audio_array),
)
elif message["event"] == "start":
if self.stream_handler.phone_mode:
self.stream_id = cast(str, message["streamSid"])
else:
self.stream_id = cast(str, message["websocket_id"])
self.set_additional_outputs = self.set_additional_outputs_factory(
self.stream_id
)
await self.set_handler(self.stream_id, self)
elif message["event"] == "stop":
self.quit.set()
self.clean_up(cast(str, self.stream_id))
return
elif message["event"] == "ping":
await websocket.send_json({"event": "pong"})
except Exception as e:
print(e)
import traceback
traceback.print_exc()
logger.debug("Error in websocket handler %s", e)
finally:
if self._emit_task:
self._emit_task.cancel()
if self._emit_to_queue_task:
self._emit_to_queue_task.cancel()
if self.start_up_task:
self.start_up_task.cancel()
await websocket.close()
async def _emit_to_queue(self):
try:
while not self.quit.is_set():
if isinstance(self.stream_handler, AsyncStreamHandler):
output = await self.stream_handler.emit()
else:
output = await run_sync(self.stream_handler.emit)
self.queue.put_nowait(output)
except asyncio.CancelledError:
logger.debug("Emit loop cancelled")
except Exception as e:
import traceback
traceback.print_exc()
logger.debug("Error in emit loop: %s", e)
async def _emit_loop(self):
try:
while not self.quit.is_set():
output = await self.queue.get()
if output is not None:
frame, output = split_output(output)
if output is not None:
self.set_additional_outputs(output)
if not isinstance(frame, tuple):
continue
target_rate = (
self.stream_handler.output_sample_rate
if not self.stream_handler.phone_mode
else 8000
)
mulaw_audio = convert_to_mulaw(
frame[1], frame[0], target_rate=target_rate
)
audio_payload = base64.b64encode(mulaw_audio).decode("utf-8")
if self.websocket and self.stream_id:
payload = {
"event": "media",
"media": {"payload": audio_payload},
}
if self.stream_handler.phone_mode:
payload["streamSid"] = self.stream_id
await self.websocket.send_json(payload)
await asyncio.sleep(0.02)
except asyncio.CancelledError:
logger.debug("Emit loop cancelled")
except Exception as e:
import traceback
traceback.print_exc()
logger.debug("Error in emit loop: %s", e)