716 lines
30 KiB
Python
716 lines
30 KiB
Python
"""将未修改的腾讯演示协议转换为 FunASR 原生在线 WebSocket 协议。"""
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from __future__ import annotations
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import argparse
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import asyncio
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import json
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import logging
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import os
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from dataclasses import dataclass
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from pathlib import Path
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from typing import Any
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from uuid import uuid4
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from aiohttp import WSMsgType, web
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from dotenv import load_dotenv
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try:
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from websockets.asyncio.client import connect as websocket_connect
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except ImportError: # websockets 13 之前的版本在包根目录提供相同客户端。
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from websockets import connect as websocket_connect
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try:
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from .auxiliary_service import AuxiliaryModelService, AuxiliaryServiceConfig
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except ImportError:
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from auxiliary_service import AuxiliaryModelService, AuxiliaryServiceConfig
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PROJECT_ROOT = Path(__file__).resolve().parents[2]
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load_dotenv(PROJECT_ROOT / ".env")
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LOGGER = logging.getLogger(__name__)
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WEB_HOST = os.getenv("WEB_HOST", "0.0.0.0")
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WEB_PORT = int(os.getenv("WEB_PORT", "8082"))
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NATIVE_WS_URL = os.getenv("FUNASR_NATIVE_WS_URL", "ws://127.0.0.1:10095")
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CHUNK_SIZE = tuple(
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int(part.strip()) for part in os.getenv("FUNASR_CHUNK_SIZE", "0,10,5").split(",")
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)
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if len(CHUNK_SIZE) != 3:
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CHUNK_SIZE = (0, 10, 5)
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CHUNK_INTERVAL = max(1, int(os.getenv("FUNASR_CHUNK_INTERVAL", "10")))
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SAMPLE_RATE = 16000
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PCM_BYTES_PER_MS = SAMPLE_RATE * 2 / 1000
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FRAME_BYTES = max(2, round(60 * CHUNK_SIZE[1] / CHUNK_INTERVAL * PCM_BYTES_PER_MS))
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MAX_SPEAKER_AUDIO_BYTES = 60 * SAMPLE_RATE * 2
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MIN_SPEAKER_AUDIO_BYTES = int(0.8 * SAMPLE_RATE * 2)
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FINALIZE_TIMEOUT_SECONDS = max(30, int(os.getenv("FUNASR_FINALIZE_TIMEOUT_SECONDS", "300")))
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TURN_SENTENCE_ID_STRIDE = 100
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AUXILIARY_KEY = web.AppKey("auxiliary_service", AuxiliaryModelService)
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SESSION_REGISTRY_KEY = web.AppKey("sessions", dict)
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class IncrementalWavDecoder:
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"""读取流式 PCM WAV 文件头，并逐段返回 16 kHz 单声道 PCM 负载。"""
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def __init__(self) -> None:
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self.buffer = bytearray()
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self.header_done = False
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self.format_valid = False
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self.data_remaining: int | None = None
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def feed(self, data: bytes) -> bytes:
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if self.header_done:
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if self.data_remaining is None:
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return data
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payload = data[: self.data_remaining]
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self.data_remaining -= len(payload)
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return payload
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self.buffer.extend(data)
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if len(self.buffer) < 12:
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return b""
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if self.buffer[:4] != b"RIFF" or self.buffer[8:12] != b"WAVE":
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raise ValueError("WAV file must use a RIFF/WAVE container")
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del self.buffer[:12]
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while len(self.buffer) >= 8:
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kind = bytes(self.buffer[:4])
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size = int.from_bytes(self.buffer[4:8], "little")
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if size > 1024 * 1024:
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raise ValueError("WAV header chunk is unexpectedly large")
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full_size = 8 + size + (size % 2)
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if len(self.buffer) < full_size:
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return b""
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body = bytes(self.buffer[8 : 8 + size])
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if kind == b"fmt ":
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if size < 16:
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raise ValueError("WAV fmt chunk is incomplete")
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fmt = (
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int.from_bytes(body[0:2], "little"),
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int.from_bytes(body[2:4], "little"),
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int.from_bytes(body[4:8], "little"),
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int.from_bytes(body[14:16], "little"),
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)
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if fmt != (1, 1, SAMPLE_RATE, 16):
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raise ValueError("WAV must be PCM16, mono, 16 kHz")
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self.format_valid = True
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if kind == b"data":
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if not self.format_valid or size % 2:
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raise ValueError("WAV must be PCM16, mono, 16 kHz")
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self.header_done = True
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self.data_remaining = size
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del self.buffer[:8]
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payload = bytes(self.buffer[:size])
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del self.buffer[: min(size, len(self.buffer))]
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self.data_remaining -= len(payload)
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return payload
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del self.buffer[:full_size]
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return b""
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def finish(self) -> None:
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if not self.header_done or self.data_remaining not in (None, 0):
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raise ValueError("WAV ended before its complete PCM data chunk arrived")
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@dataclass(frozen=True)
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class SpeakerJob:
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sentence_id: int
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text: str
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audio: bytes
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start_time_ms: float
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end_time_ms: float
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def split_text_by_speaker_segments(
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text: str,
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segments: list[dict[str, Any]],
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turn_start_ms: float,
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turn_end_ms: float,
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) -> list[dict[str, Any]]:
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"""按 CAM++ 时间段近似拆分 ASR 文本，并保留每段稳定说话人标签。"""
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usable: list[dict[str, Any]] = []
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for segment in sorted(
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segments,
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key=lambda item: float(item.get("start_time", turn_start_ms)),
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):
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try:
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start = max(turn_start_ms, float(segment.get("start_time", turn_start_ms)))
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end = min(turn_end_ms, float(segment.get("end_time", turn_end_ms)))
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speaker_id = int(segment.get("speaker_id", -1))
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except (TypeError, ValueError):
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continue
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if end <= start or speaker_id < 0:
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continue
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speaker = {
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"speaker_id": speaker_id,
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"speaker_name": str(segment.get("speaker_name") or f"说话人 {speaker_id + 1}"),
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"speaker_confidence": float(segment.get("speaker_confidence") or 0),
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"speaker_status": str(segment.get("speaker_status") or "confirmed"),
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}
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if usable and usable[-1]["speaker"]["speaker_id"] == speaker_id:
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usable[-1]["end_time_ms"] = end
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else:
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usable.append(
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{"start_time_ms": start, "end_time_ms": end, "speaker": speaker}
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)
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if not usable:
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return []
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if len(usable) == 1:
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usable[0]["text"] = text
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return usable
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# 双重保护：服务端有最短段长滤波，桥接层也拒绝意外的短片段响应。
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min_split_ms = max(1500, int(os.getenv("FUNASR_SPEAKER_MIN_SEGMENT_MS", "3000")))
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while len(usable) > 1:
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short_index = next(
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(
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index
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for index, item in enumerate(usable)
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if item["end_time_ms"] - item["start_time_ms"] < min_split_ms
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),
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None,
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)
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if short_index is None:
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break
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if short_index == 0:
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usable[1]["start_time_ms"] = usable[0]["start_time_ms"]
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del usable[0]
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elif short_index == len(usable) - 1:
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usable[-2]["end_time_ms"] = usable[-1]["end_time_ms"]
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del usable[-1]
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else:
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previous = usable[short_index - 1]
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following = usable[short_index + 1]
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if previous["end_time_ms"] - previous["start_time_ms"] >= (
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following["end_time_ms"] - following["start_time_ms"]
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):
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previous["end_time_ms"] = usable[short_index]["end_time_ms"]
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del usable[short_index]
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else:
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following["start_time_ms"] = usable[short_index]["start_time_ms"]
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del usable[short_index]
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if len(usable) == 1 or len(text) < len(usable):
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usable = [max(usable, key=lambda item: item["end_time_ms"] - item["start_time_ms"])]
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usable[0]["text"] = text
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return usable
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total_duration = sum(item["end_time_ms"] - item["start_time_ms"] for item in usable)
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char_start = 0
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elapsed = 0.0
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for index, item in enumerate(usable):
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elapsed += item["end_time_ms"] - item["start_time_ms"]
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if index == len(usable) - 1:
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char_end = len(text)
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else:
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char_end = round(len(text) * elapsed / total_duration)
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char_end = max(char_start + 1, min(char_end, len(text) - (len(usable) - index - 1)))
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item["text"] = text[char_start:char_end].strip()
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char_start = char_end
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return [item for item in usable if item.get("text")]
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class BrowserSession:
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"""管理浏览器与原生 WebSocket 连接，并转换双方的消息格式。"""
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def __init__(
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self,
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browser_ws: web.WebSocketResponse,
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auxiliary: AuxiliaryModelService,
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start: dict[str, Any],
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voice_id: str,
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) -> None:
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self.browser_ws = browser_ws
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self.auxiliary = auxiliary
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self.start = start
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self.voice_id = voice_id
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self.session_id = uuid4().hex
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self.stop_event = asyncio.Event()
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self.send_lock = asyncio.Lock()
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self.speaker_jobs: asyncio.Queue[SpeakerJob | None] = asyncio.Queue()
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self.wav_decoder = (
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IncrementalWavDecoder()
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if str(start.get("source") or "mic") == "file"
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and Path(str(start.get("file_name") or "")).suffix.lower() == ".wav"
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else None
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)
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self.speed_factor = max(0.5, min(3.0, float(start.get("speed_factor") or 1.0)))
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try:
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sentence_strategy = int(start.get("sentence_strategy", 0))
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except (TypeError, ValueError):
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sentence_strategy = 0
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self.sentence_strategy = sentence_strategy if sentence_strategy in (0, 1) else 0
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self.pending_pcm = bytearray()
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self.turn_audio = bytearray()
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self.total_audio_ms = 0.0
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self.turn_start_ms = 0.0
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self.turn_text = ""
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self.sentence_id = 0
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self.native_ack: dict[str, Any] | None = None
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self.native_error: str | None = None
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async def emit(self, payload: dict[str, Any]) -> None:
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"""ASR 和 CAM++ 的完成时机不同，因此需要串行写入浏览器连接。"""
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async with self.send_lock:
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if not self.browser_ws.closed:
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await self.browser_ws.send_json(payload)
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async def emit_sentence(
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self,
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text: str,
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final: bool,
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speaker: dict[str, Any] | None = None,
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sentence_id: int | None = None,
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start_time_ms: float | None = None,
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end_time_ms: float | None = None,
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) -> None:
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speaker = speaker or {}
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sentence = {
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"sentence_id": self.sentence_id if sentence_id is None else sentence_id,
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"sentence": text,
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"sentence_type": 1 if final else 0,
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"start_time": round(self.turn_start_ms if start_time_ms is None else start_time_ms),
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"end_time": round(self.total_audio_ms if end_time_ms is None else end_time_ms),
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# 未修改的腾讯界面使用 speaker_id 选择对应的说话人气泡。
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"speaker_id": int(speaker.get("speaker_id", -1)),
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"speaker_name": str(speaker.get("speaker_name") or ""),
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"speaker_confidence": float(speaker.get("speaker_confidence") or 0),
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"speaker_status": str(speaker.get("speaker_status") or "pending"),
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}
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await self.emit({"type": "sentences", "sentences": [sentence]})
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def align_turn_audio_to_vad(self, message: dict[str, Any]) -> None:
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"""用原生 VAD 的绝对时间裁掉当前 turn 的前后静音样本。"""
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buffer_end_ms = self.turn_start_ms + len(self.turn_audio) / PCM_BYTES_PER_MS
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try:
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speech_start_ms = message.get("speech_start_ms")
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if speech_start_ms is not None:
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target = max(self.turn_start_ms, min(buffer_end_ms, float(speech_start_ms)))
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trim_bytes = int(round((target - self.turn_start_ms) * PCM_BYTES_PER_MS))
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trim_bytes -= trim_bytes % 2
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del self.turn_audio[:trim_bytes]
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self.turn_start_ms += trim_bytes / PCM_BYTES_PER_MS
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buffer_end_ms = self.turn_start_ms + len(self.turn_audio) / PCM_BYTES_PER_MS
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speech_end_ms = message.get("speech_end_ms")
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if speech_end_ms is not None:
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target = max(self.turn_start_ms, min(buffer_end_ms, float(speech_end_ms)))
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keep_bytes = int(round((target - self.turn_start_ms) * PCM_BYTES_PER_MS))
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keep_bytes -= keep_bytes % 2
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del self.turn_audio[keep_bytes:]
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except (TypeError, ValueError):
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LOGGER.warning("Ignoring invalid native FunASR VAD boundary: %s", message)
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async def send_pcm_frame(self, native_ws: Any, frame: bytes, pace_file: bool) -> None:
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if not frame:
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return
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if len(frame) % 2:
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raise ValueError("PCM16 audio ended on an incomplete sample")
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self.total_audio_ms += len(frame) / PCM_BYTES_PER_MS
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self.turn_audio.extend(frame)
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if len(self.turn_audio) > MAX_SPEAKER_AUDIO_BYTES:
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# 即使 VAD 始终没有报告端点，也要限制每轮音频的内存占用。
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trim = len(self.turn_audio) - MAX_SPEAKER_AUDIO_BYTES
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del self.turn_audio[:trim]
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self.turn_start_ms += trim / PCM_BYTES_PER_MS
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await native_ws.send(frame)
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if pace_file:
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await asyncio.sleep(len(frame) / (SAMPLE_RATE * 2) / self.speed_factor)
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async def accept_audio(self, native_ws: Any, data: bytes) -> None:
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pcm = self.wav_decoder.feed(data) if self.wav_decoder else data
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if not pcm:
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return
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self.pending_pcm.extend(pcm)
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is_file = str(self.start.get("source") or "mic") == "file"
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while len(self.pending_pcm) >= FRAME_BYTES:
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frame = bytes(self.pending_pcm[:FRAME_BYTES])
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del self.pending_pcm[:FRAME_BYTES]
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await self.send_pcm_frame(native_ws, frame, is_file)
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async def finish_audio(self, native_ws: Any) -> None:
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if self.wav_decoder:
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self.wav_decoder.finish()
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if self.pending_pcm:
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frame = bytes(self.pending_pcm)
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self.pending_pcm.clear()
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await self.send_pcm_frame(
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native_ws, frame, str(self.start.get("source") or "mic") == "file"
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)
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async def read_native(self, native_ws: Any) -> None:
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"""接收 FunASR 原生事件，并维护每段话语的中间结果缓存。"""
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try:
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while True:
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raw = await native_ws.recv()
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message = json.loads(raw)
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if message.get("is_end"):
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self.native_ack = message
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return
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if message.get("event") == "vad":
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self.align_turn_audio_to_vad(message)
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continue
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text = str(message.get("text") or "")
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if text:
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# FunASR 在线模式会为每个分块发送新解码出的文本。
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self.turn_text += text
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if message.get("is_final"):
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final_text = self.turn_text.strip()
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audio = bytes(self.turn_audio)
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start_ms = self.turn_start_ms
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end_ms = start_ms + len(audio) / PCM_BYTES_PER_MS
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turn_sentence_id = self.sentence_id
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if final_text:
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# 先结束前端中间结果气泡；标点和 CAM++ 在独立工作线程中完成，
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# 不阻塞原生 WebSocket 继续读取后续音频帧。
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await self.emit_sentence(
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final_text,
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final=True,
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sentence_id=turn_sentence_id,
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start_time_ms=start_ms,
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end_time_ms=end_ms,
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)
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await self.speaker_jobs.put(
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SpeakerJob(
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sentence_id=turn_sentence_id,
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text=final_text,
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audio=audio,
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start_time_ms=start_ms,
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end_time_ms=end_ms,
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)
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)
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# 为同一个 VAD 轮次内可能拆出的多个气泡预留独立 ID。
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self.sentence_id += TURN_SENTENCE_ID_STRIDE
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self.turn_text = ""
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self.turn_audio.clear()
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self.turn_start_ms = self.total_audio_ms
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elif self.turn_text:
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await self.emit_sentence(self.turn_text, final=False)
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except asyncio.CancelledError:
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raise
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except Exception as exc:
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self.native_error = str(exc)
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LOGGER.exception("native FunASR WebSocket closed unexpectedly")
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await self.emit({"type": "error", "message": f"FunASR realtime WS: {exc}"})
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async def resolve_speakers(self) -> None:
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"""按序定稿标点并用 CAM++ 滑窗恢复 turn 内说话人切换。"""
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while True:
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job = await self.speaker_jobs.get()
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try:
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if job is None:
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return
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final_text = job.text
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try:
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punctuation = await self.auxiliary.punctuate(final_text)
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if not punctuation.get("available", False):
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LOGGER.warning(
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"FunASR punctuation is unavailable: %s",
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punctuation.get("error", "model is not configured"),
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)
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else:
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punctuated = str(punctuation.get("text") or "").strip()
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if punctuated:
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final_text = punctuated
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except Exception:
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LOGGER.exception("FunASR punctuation request failed; keeping raw text")
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subsegments: list[dict[str, Any]] = []
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if len(job.audio) >= MIN_SPEAKER_AUDIO_BYTES:
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# 必需的 CAM++ 按 FunASR 的 1.5 秒窗口和 0.75 秒步长，
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# 在同一个 VAD 轮次内识别多个说话人片段。
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track = getattr(self.auxiliary, "track_speakers", None)
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if callable(track):
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try:
|
|
tracked = await track(
|
|
job.audio,
|
|
self.session_id,
|
|
job.start_time_ms,
|
|
job.end_time_ms,
|
|
)
|
|
subsegments = split_text_by_speaker_segments(
|
|
final_text,
|
|
tracked,
|
|
job.start_time_ms,
|
|
job.end_time_ms,
|
|
)
|
|
except Exception:
|
|
LOGGER.exception(
|
|
"CAM++ sliding-window diarization failed; falling back to whole-turn speaker"
|
|
)
|
|
|
|
if not subsegments and len(job.audio) >= MIN_SPEAKER_AUDIO_BYTES:
|
|
try:
|
|
resolved = await self.auxiliary.resolve_speaker(
|
|
job.audio,
|
|
self.session_id,
|
|
job.start_time_ms,
|
|
job.end_time_ms,
|
|
)
|
|
if resolved and int(resolved.get("speaker_id", -1)) >= 0:
|
|
subsegments = [
|
|
{
|
|
"text": final_text,
|
|
"speaker": resolved,
|
|
"start_time_ms": job.start_time_ms,
|
|
"end_time_ms": job.end_time_ms,
|
|
}
|
|
]
|
|
except Exception:
|
|
LOGGER.exception("CAM++ speaker resolution failed: voice_id=%s", self.voice_id)
|
|
|
|
if not subsegments:
|
|
subsegments = [
|
|
{
|
|
"text": final_text,
|
|
"speaker": {"speaker_id": -1},
|
|
"start_time_ms": job.start_time_ms,
|
|
"end_time_ms": job.end_time_ms,
|
|
}
|
|
]
|
|
|
|
for index, segment in enumerate(subsegments):
|
|
await self.emit_sentence(
|
|
str(segment["text"]),
|
|
final=True,
|
|
speaker=segment.get("speaker"),
|
|
sentence_id=job.sentence_id + index,
|
|
start_time_ms=float(segment.get("start_time_ms", job.start_time_ms)),
|
|
end_time_ms=float(segment.get("end_time_ms", job.end_time_ms)),
|
|
)
|
|
finally:
|
|
self.speaker_jobs.task_done()
|
|
|
|
|
|
async def config_handler(_: web.Request) -> web.Response:
|
|
"""提供简洁的就绪状态响应，供启动器和诊断使用。"""
|
|
return web.json_response(
|
|
{
|
|
"engine": "funasr-native-online-ws",
|
|
"model": os.getenv("FUNASR_ASR_MODEL", ""),
|
|
"native_ws_url": NATIVE_WS_URL,
|
|
"speaker_service_url": os.getenv(
|
|
"AUXILIARY_SERVICE_URL", "http://127.0.0.1:8010"
|
|
),
|
|
}
|
|
)
|
|
|
|
|
|
async def stop_handler(request: web.Request) -> web.Response:
|
|
voice_id = request.query.get("voice_id", "").strip()
|
|
if not voice_id:
|
|
return web.json_response({"ok": False, "error": "missing voice_id"}, status=400)
|
|
stop_event = request.app[SESSION_REGISTRY_KEY].get(voice_id)
|
|
if stop_event is None:
|
|
return web.json_response({"ok": False, "error": "session not found"})
|
|
stop_event.set()
|
|
return web.json_response({"ok": True})
|
|
|
|
|
|
async def websocket_handler(request: web.Request) -> web.WebSocketResponse:
|
|
"""将腾讯前端消息桥接到 FunASR 原生实时协议。"""
|
|
browser_ws = web.WebSocketResponse(max_msg_size=64 * 1024 * 1024, heartbeat=30)
|
|
await browser_ws.prepare(request)
|
|
session: BrowserSession | None = None
|
|
native_reader: asyncio.Task[None] | None = None
|
|
speaker_worker: asyncio.Task[None] | None = None
|
|
voice_id = ""
|
|
registered = False
|
|
|
|
try:
|
|
first = await browser_ws.receive()
|
|
if first.type != WSMsgType.TEXT:
|
|
await browser_ws.send_json({"type": "error", "message": "first message must be JSON start"})
|
|
return browser_ws
|
|
start = json.loads(first.data)
|
|
if not isinstance(start, dict) or start.get("type") != "start":
|
|
await browser_ws.send_json({"type": "error", "message": "first message must have type=start"})
|
|
return browser_ws
|
|
source = str(start.get("source") or "mic")
|
|
suffix = Path(str(start.get("file_name") or "")).suffix.lower()
|
|
if source == "file" and suffix not in {".pcm", ".wav"}:
|
|
await browser_ws.send_json(
|
|
{
|
|
"type": "error",
|
|
"message": "FunASR realtime mode accepts PCM or 16 kHz mono PCM WAV files.",
|
|
}
|
|
)
|
|
return browser_ws
|
|
if source == "file" and suffix == ".pcm":
|
|
start = dict(start)
|
|
start["file_name"] = str(start.get("file_name") or "audio.pcm")
|
|
|
|
voice_id = uuid4().hex
|
|
auxiliary: AuxiliaryModelService = request.app[AUXILIARY_KEY]
|
|
session = BrowserSession(browser_ws, auxiliary, start, voice_id)
|
|
request.app[SESSION_REGISTRY_KEY][voice_id] = session.stop_event
|
|
registered = True
|
|
await session.emit({"type": "voice_id", "voice_id": voice_id})
|
|
await session.emit({"type": "start"})
|
|
|
|
# 这是 FunASR 原生 WSS 消息格式；之后每 60 毫秒发送一帧 PCM。
|
|
async with websocket_connect(
|
|
NATIVE_WS_URL,
|
|
subprotocols=["binary"],
|
|
ping_interval=None,
|
|
close_timeout=3,
|
|
max_size=None,
|
|
) as native_ws:
|
|
await native_ws.send(
|
|
json.dumps(
|
|
{
|
|
"mode": "online",
|
|
"chunk_size": list(CHUNK_SIZE),
|
|
"chunk_interval": CHUNK_INTERVAL,
|
|
"encoder_chunk_look_back": int(
|
|
os.getenv("FUNASR_ENCODER_LOOK_BACK", "4")
|
|
),
|
|
"decoder_chunk_look_back": int(
|
|
os.getenv("FUNASR_DECODER_LOOK_BACK", "1")
|
|
),
|
|
"sentence_strategy": session.sentence_strategy,
|
|
"audio_fs": SAMPLE_RATE,
|
|
"wav_name": voice_id,
|
|
"is_speaking": True,
|
|
},
|
|
ensure_ascii=False,
|
|
)
|
|
)
|
|
native_reader = asyncio.create_task(session.read_native(native_ws))
|
|
speaker_worker = asyncio.create_task(session.resolve_speakers())
|
|
|
|
while not browser_ws.closed:
|
|
receive_task = asyncio.create_task(browser_ws.receive())
|
|
stop_task = asyncio.create_task(session.stop_event.wait())
|
|
done, _ = await asyncio.wait(
|
|
[receive_task, stop_task, native_reader],
|
|
return_when=asyncio.FIRST_COMPLETED,
|
|
)
|
|
if stop_task in done:
|
|
receive_task.cancel()
|
|
await asyncio.gather(receive_task, return_exceptions=True)
|
|
break
|
|
stop_task.cancel()
|
|
await asyncio.gather(stop_task, return_exceptions=True)
|
|
if native_reader in done:
|
|
receive_task.cancel()
|
|
await asyncio.gather(receive_task, return_exceptions=True)
|
|
if session.native_ack is None and session.native_error is None:
|
|
session.native_error = "FunASR native WebSocket ended before EOF acknowledgement"
|
|
break
|
|
|
|
message = await receive_task
|
|
if message.type == WSMsgType.BINARY:
|
|
await session.accept_audio(native_ws, bytes(message.data))
|
|
continue
|
|
if message.type == WSMsgType.TEXT:
|
|
try:
|
|
control = json.loads(message.data)
|
|
except json.JSONDecodeError:
|
|
continue
|
|
if isinstance(control, dict) and control.get("type") in {"eof", "stop"}:
|
|
break
|
|
if isinstance(control, dict) and control.get("type") == "abort":
|
|
session.native_error = "session aborted by browser"
|
|
break
|
|
if message.type in {WSMsgType.CLOSE, WSMsgType.CLOSED, WSMsgType.ERROR}:
|
|
break
|
|
|
|
if session.native_error is None and not browser_ws.closed:
|
|
await session.finish_audio(native_ws)
|
|
# FunASR 会刷新在线缓存，并在输出最终结果后才返回确认。
|
|
await native_ws.send(
|
|
json.dumps({"is_speaking": False, "is_end": True}, ensure_ascii=False)
|
|
)
|
|
try:
|
|
await asyncio.wait_for(native_reader, timeout=FINALIZE_TIMEOUT_SECONDS)
|
|
except asyncio.TimeoutError:
|
|
session.native_error = (
|
|
f"FunASR did not acknowledge end-of-input within "
|
|
f"{FINALIZE_TIMEOUT_SECONDS}s"
|
|
)
|
|
|
|
if speaker_worker is not None:
|
|
await session.speaker_jobs.join()
|
|
await session.speaker_jobs.put(None)
|
|
await speaker_worker
|
|
speaker_worker = None
|
|
if session.native_error:
|
|
await session.emit({"type": "error", "message": session.native_error})
|
|
elif session.native_ack and not session.native_ack.get("is_final", False):
|
|
await session.emit(
|
|
{
|
|
"type": "error",
|
|
"message": str(session.native_ack.get("error") or "FunASR did not finalize the stream"),
|
|
}
|
|
)
|
|
elif session.native_ack:
|
|
await session.emit({"type": "end"})
|
|
except asyncio.CancelledError:
|
|
raise
|
|
except Exception as exc:
|
|
LOGGER.exception("Tencent-compatible WebSocket session failed")
|
|
if not browser_ws.closed:
|
|
await browser_ws.send_json({"type": "error", "message": str(exc)})
|
|
finally:
|
|
if native_reader is not None and not native_reader.done():
|
|
native_reader.cancel()
|
|
await asyncio.gather(native_reader, return_exceptions=True)
|
|
if speaker_worker is not None and not speaker_worker.done():
|
|
speaker_worker.cancel()
|
|
await asyncio.gather(speaker_worker, return_exceptions=True)
|
|
if registered:
|
|
request.app[SESSION_REGISTRY_KEY].pop(voice_id, None)
|
|
if session is not None:
|
|
await session.auxiliary.reset_speaker_session(session.session_id)
|
|
if not browser_ws.closed:
|
|
await browser_ws.close()
|
|
return browser_ws
|
|
|
|
|
|
async def create_app() -> web.Application:
|
|
"""创建轻量协议桥接层；模型推理由 FunASR 原生服务负责。"""
|
|
app = web.Application()
|
|
app[AUXILIARY_KEY] = AuxiliaryModelService(
|
|
AuxiliaryServiceConfig(
|
|
base_url=os.getenv("AUXILIARY_SERVICE_URL", "http://127.0.0.1:8010")
|
|
)
|
|
)
|
|
app[SESSION_REGISTRY_KEY] = {}
|
|
|
|
async def lifecycle(application: web.Application):
|
|
await application[AUXILIARY_KEY].start()
|
|
try:
|
|
health = await application[AUXILIARY_KEY].health()
|
|
if not health.get("speaker_embedding_ready"):
|
|
raise RuntimeError("CAM++ speaker service is not ready")
|
|
except Exception:
|
|
await application[AUXILIARY_KEY].close()
|
|
raise
|
|
yield
|
|
await application[AUXILIARY_KEY].close()
|
|
|
|
app.cleanup_ctx.append(lifecycle)
|
|
app.router.add_get("/api/config", config_handler)
|
|
app.router.add_get("/api/stop", stop_handler)
|
|
app.router.add_get("/ws", websocket_handler)
|
|
return app
|
|
|
|
|
|
def main() -> None:
|
|
parser = argparse.ArgumentParser(description=__doc__)
|
|
parser.add_argument("--no-browser", action="store_true")
|
|
args = parser.parse_args()
|
|
logging.basicConfig(level=logging.INFO)
|
|
print(
|
|
f"FunASR browser bridge: http://{os.getenv('WEB_DISPLAY_HOST', '127.0.0.1')}:{WEB_PORT}/api/config",
|
|
flush=True,
|
|
)
|
|
web.run_app(create_app(), host=WEB_HOST, port=WEB_PORT)
|
|
|
|
|
|
if __name__ == "__main__":
|
|
main()
|