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https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-22 07:03:53 +08:00
update:添加了阿里云CosyVoice流式TTS
This commit is contained in:
CGD
@@ -690,6 +690,28 @@ TTS:
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# pitch_rate: 0
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# 添加 302.ai TTS 配置
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# token申请地址：https://dash.302.ai/
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AliyunStreamTTS:
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# 阿里云CosyVoice大模型流式文本语音合成
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# 采用FlowingSpeechSynthesizer接口，支持更低延迟和更自然的语音质量
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# 流式文本语音合成仅提供商用版，不支持试用，详情请参见试用版和商用版。要使用该功能，请开通商用版。
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# 支持龙系列专用音色：longxiaochun、longyu、longchen等
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# 平台地址：https://nls-portal.console.aliyun.com/
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# appkey地址：https://nls-portal.console.aliyun.com/applist
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# token地址：https://nls-portal.console.aliyun.com/overview
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# 使用三阶段流式交互：StartSynthesis -> RunSynthesis -> StopSynthesis
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type: aliyun
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output_dir: tmp/
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appkey: 你的阿里云智能语音交互服务项目Appkey
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token: 你的阿里云智能语音交互服务AccessToken，临时的24小时，要长期用下方的access_key_id，access_key_secret
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voice: longxiaochun
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access_key_id: 你的阿里云账号access_key_id
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access_key_secret: 你的阿里云账号access_key_secret
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# 以下可不用设置，使用默认设置
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# format: pcm # 音频格式：pcm、wav、mp3
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# sample_rate: 16000 # 采样率：8000、16000、24000
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# volume: 50 # 音量：0-100
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# speech_rate: 0 # 语速：-500到500
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# pitch_rate: 0 # 语调：-500到500
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TencentTTS:
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# 腾讯云智能语音交互服务，需要先在腾讯云平台开通服务
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# appid、secret_id、secret_key申请地址：https://console.cloud.tencent.com/cam/capi
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@@ -0,0 +1,891 @@
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import uuid
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import json
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import hmac
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import hashlib
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import base64
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import time
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import queue
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import asyncio
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import traceback
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import websockets
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import websockets.protocol
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import os
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import random
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import threading
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import concurrent.futures
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import sys
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from datetime import datetime
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from urllib import parse
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from typing import Optional
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from core.providers.tts.base import TTSProviderBase
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from core.providers.tts.dto.dto import SentenceType, ContentType, InterfaceType
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from core.utils.tts import MarkdownCleaner
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from core.utils import opus_encoder_utils, textUtils
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from config.logger import setup_logging
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TAG = __name__
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logger = setup_logging()
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class AccessToken:
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@staticmethod
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def _encode_text(text):
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encoded_text = parse.quote_plus(text)
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return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
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@staticmethod
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def _encode_dict(dic):
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keys = dic.keys()
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dic_sorted = [(key, dic[key]) for key in sorted(keys)]
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encoded_text = parse.urlencode(dic_sorted)
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return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
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@staticmethod
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def create_token(access_key_id, access_key_secret):
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parameters = {
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"AccessKeyId": access_key_id,
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"Action": "CreateToken",
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"Format": "JSON",
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"RegionId": "cn-shanghai", # 使用上海地域进行Token获取
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"SignatureMethod": "HMAC-SHA1",
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"SignatureNonce": str(uuid.uuid1()),
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"SignatureVersion": "1.0",
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"Timestamp": time.strftime("%Y-%m-%dT%H:%M:%SZ", time.gmtime()),
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"Version": "2019-02-28",
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}
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query_string = AccessToken._encode_dict(parameters)
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string_to_sign = (
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"GET"
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+ "&"
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+ AccessToken._encode_text("/")
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+ "&"
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+ AccessToken._encode_text(query_string)
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)
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secreted_string = hmac.new(
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bytes(access_key_secret + "&", encoding="utf-8"),
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bytes(string_to_sign, encoding="utf-8"),
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hashlib.sha1,
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).digest()
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signature = base64.b64encode(secreted_string)
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signature = AccessToken._encode_text(signature)
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full_url = "http://nls-meta.cn-shanghai.aliyuncs.com/?Signature=%s&%s" % (
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signature,
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query_string,
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)
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import requests
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response = requests.get(full_url)
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if response.ok:
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root_obj = response.json()
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key = "Token"
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if key in root_obj:
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token = root_obj[key]["Id"]
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expire_time = root_obj[key]["ExpireTime"]
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return token, expire_time
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return None, None
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class TTSProvider(TTSProviderBase):
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def __init__(self, config, delete_audio_file):
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super().__init__(config, delete_audio_file)
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# 设置为流式接口类型
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self.interface_type = InterfaceType.SINGLE_STREAM
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# 基础配置
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self.access_key_id = config.get("access_key_id")
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self.access_key_secret = config.get("access_key_secret")
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self.appkey = config.get("appkey")
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self.format = config.get("format", "pcm")
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self.audio_file_type = config.get("format", "pcm")
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# 采样率配置
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sample_rate = config.get("sample_rate", "16000")
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self.sample_rate = int(sample_rate) if sample_rate else 16000
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# 音色配置 - CosyVoice大模型音色
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if config.get("private_voice"):
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self.voice = config.get("private_voice")
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else:
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self.voice = config.get("voice", "longxiaochun") # CosyVoice默认音色
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# 音频参数配置
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volume = config.get("volume", "50")
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self.volume = int(volume) if volume else 50
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speech_rate = config.get("speech_rate", "0")
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self.speech_rate = int(speech_rate) if speech_rate else 0
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pitch_rate = config.get("pitch_rate", "0")
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self.pitch_rate = int(pitch_rate) if pitch_rate else 0
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# WebSocket配置
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self.host = config.get("host", "nls-gateway-cn-beijing.aliyuncs.com")
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self.ws_url = f"wss://{self.host}/ws/v1"
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self.ws = None
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# 流式相关配置
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self.before_stop_play_files = []
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self.segment_count = 0
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# 创建Opus编码器
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self.opus_encoder = opus_encoder_utils.OpusEncoderUtils(
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sample_rate=16000, channels=1, frame_size_ms=60
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)
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# PCM缓冲区
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self.pcm_buffer = bytearray()
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# Token管理
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if self.access_key_id and self.access_key_secret:
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self._refresh_token()
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else:
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self.token = config.get("token")
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self.expire_time = None
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def _refresh_token(self):
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"""刷新Token并记录过期时间"""
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if self.access_key_id and self.access_key_secret:
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self.token, expire_time_str = AccessToken.create_token(
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self.access_key_id, self.access_key_secret
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)
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if not expire_time_str:
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raise ValueError("无法获取有效的Token过期时间")
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try:
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expire_str = str(expire_time_str).strip()
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if expire_str.isdigit():
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expire_time = datetime.fromtimestamp(int(expire_str))
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else:
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expire_time = datetime.strptime(expire_str, "%Y-%m-%dT%H:%M:%SZ")
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self.expire_time = expire_time.timestamp() - 60
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except Exception as e:
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raise ValueError(f"无效的过期时间格式: {expire_str}") from e
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else:
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self.expire_time = None
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if not self.token:
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raise ValueError("无法获取有效的访问Token")
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def _is_token_expired(self):
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"""检查Token是否过期"""
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if not self.expire_time:
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return False
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return time.time() > self.expire_time
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async def _ensure_connection(self):
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"""确保WebSocket连接可用"""
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# 检查连接状态，兼容不同版本的websockets库
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need_reconnect = False
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if self.ws is None:
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need_reconnect = True
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else:
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try:
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# 尝试访问closed属性，如果不存在则检查state
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if hasattr(self.ws, 'closed'):
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need_reconnect = self.ws.closed
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elif hasattr(self.ws, 'state'):
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# websockets 新版本使用state属性
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need_reconnect = self.ws.state != websockets.protocol.State.OPEN
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else:
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# 如果都没有，尝试发送ping来检测连接状态
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try:
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await asyncio.wait_for(self.ws.ping(), timeout=2.0)
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except:
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need_reconnect = True
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except:
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need_reconnect = True
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if need_reconnect:
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# 清理旧连接
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if self.ws:
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try:
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if hasattr(self.ws, 'close'):
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if asyncio.iscoroutinefunction(self.ws.close):
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await self.ws.close()
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else:
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self.ws.close()
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except:
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pass
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finally:
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self.ws = None
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if self._is_token_expired():
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logger.bind(tag=TAG).warning("Token已过期，正在自动刷新...")
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self._refresh_token()
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# 重试连接机制
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max_retries = 3
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retry_delay = 1.0
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for attempt in range(max_retries):
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try:
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self.ws = await asyncio.wait_for(
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websockets.connect(
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self.ws_url,
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additional_headers={
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"X-NLS-Token": self.token,
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},
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ping_interval=30,
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ping_timeout=10,
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close_timeout=10,
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),
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timeout=10.0
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)
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logger.bind(tag=TAG).info("阿里云CosyVoice流式TTS WebSocket连接建立成功")
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return
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except Exception as e:
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logger.bind(tag=TAG).warning(f"WebSocket连接失败 (尝试 {attempt + 1}/{max_retries}): {e}")
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if attempt < max_retries - 1:
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await asyncio.sleep(retry_delay * (attempt + 1))
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else:
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logger.bind(tag=TAG).error(f"WebSocket连接最终失败: {e}")
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raise
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def tts_text_priority_thread(self):
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"""流式文本处理线程"""
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while not self.conn.stop_event.is_set():
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try:
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message = self.tts_text_queue.get(timeout=1)
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if message.sentence_type == SentenceType.FIRST:
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# 初始化参数
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self.tts_stop_request = False
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self.processed_chars = 0
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self.tts_text_buff = []
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self.segment_count = 0
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self.tts_audio_first_sentence = True
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self.before_stop_play_files.clear()
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elif ContentType.TEXT == message.content_type:
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self.tts_text_buff.append(message.content_detail)
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segment_text = self._get_segment_text()
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if segment_text:
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self.to_tts_single_stream(segment_text)
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elif ContentType.FILE == message.content_type:
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logger.bind(tag=TAG).info(
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f"添加音频文件到待播放列表: {message.content_file}"
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)
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self.before_stop_play_files.append(
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(message.content_file, message.content_detail)
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)
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if message.sentence_type == SentenceType.LAST:
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# 处理剩余的文本
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self._process_remaining_text(True)
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except queue.Empty:
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continue
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except Exception as e:
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logger.bind(tag=TAG).error(
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f"处理TTS文本失败: {str(e)}, 类型: {type(e).__name__}, 堆栈: {traceback.format_exc()}"
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)
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def _process_remaining_text(self, is_last=False):
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"""处理剩余的文本并生成语音"""
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full_text = "".join(self.tts_text_buff)
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remaining_text = full_text[self.processed_chars:]
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if remaining_text:
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segment_text = textUtils.get_string_no_punctuation_or_emoji(remaining_text)
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if segment_text:
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self.to_tts_single_stream(segment_text, is_last)
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self.processed_chars += len(full_text)
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else:
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self._process_before_stop_play_files()
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else:
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self._process_before_stop_play_files()
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def to_tts_single_stream(self, text, is_last=False):
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"""流式TTS处理 - 使用线程池执行异步任务"""
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try:
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text = MarkdownCleaner.clean_markdown(text)
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# 使用线程池来执行异步任务，避免事件循环冲突
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def run_async_task():
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"""在新线程中运行异步任务"""
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try:
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# 创建新的事件循环用于这个线程
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loop = asyncio.new_event_loop()
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asyncio.set_event_loop(loop)
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# Windows下设置事件循环策略
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if sys.platform == "win32":
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asyncio.set_event_loop_policy(asyncio.WindowsProactorEventLoopPolicy())
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try:
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# 运行TTS任务
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loop.run_until_complete(self._tts_request_unified(text, is_last))
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return True
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finally:
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# 安全关闭事件循环
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try:
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# 取消所有未完成的任务
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pending = asyncio.all_tasks(loop)
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if pending:
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for task in pending:
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task.cancel()
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# 等待任务取消完成
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loop.run_until_complete(asyncio.gather(*pending, return_exceptions=True))
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except Exception as cleanup_error:
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logger.bind(tag=TAG).debug(f"清理事件循环异常: {cleanup_error}")
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finally:
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loop.close()
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except Exception as e:
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logger.bind(tag=TAG).error(f"异步任务执行失败: {e}")
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return False
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# 使用线程池执行异步任务
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with concurrent.futures.ThreadPoolExecutor(max_workers=1) as executor:
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future = executor.submit(run_async_task)
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success = future.result(timeout=30) # 30秒超时
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if success:
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logger.bind(tag=TAG).info(f"语音生成成功: {text}")
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else:
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logger.bind(tag=TAG).error(f"语音生成失败: {text}")
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self.tts_audio_queue.put((SentenceType.LAST, [], None))
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except concurrent.futures.TimeoutError:
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logger.bind(tag=TAG).error(f"TTS任务超时: {text}")
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self.tts_audio_queue.put((SentenceType.LAST, [], None))
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except Exception as e:
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logger.bind(tag=TAG).error(f"TTS处理异常: {text}, 错误: {e}")
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self.tts_audio_queue.put((SentenceType.LAST, [], None))
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return None
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async def text_to_speak(self, text, is_last=False):
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"""流式处理TTS音频"""
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try:
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# 确保连接可用
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await self._ensure_connection()
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ws_connection = self.ws
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# 确保Token有效
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if self._is_token_expired():
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self._refresh_token()
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# 生成task_id和message_id
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task_id = str(uuid.uuid4()).replace('-', '')
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# 第一阶段：发送StartSynthesis指令（设置参数）
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start_message_id = str(uuid.uuid4()).replace('-', '')
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start_request = {
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"header": {
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"message_id": start_message_id,
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"task_id": task_id,
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"namespace": "FlowingSpeechSynthesizer",
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"name": "StartSynthesis",
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"appkey": self.appkey,
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},
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"payload": {
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"voice": self.voice,
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"format": self.format,
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"sample_rate": self.sample_rate,
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"volume": self.volume,
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"speech_rate": self.speech_rate,
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"pitch_rate": self.pitch_rate,
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}
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}
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await ws_connection.send(json.dumps(start_request))
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logger.bind(tag=TAG).debug(f"发送StartSynthesis指令: {start_message_id}")
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# 第二阶段：发送RunSynthesis指令（发送文本）
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run_message_id = str(uuid.uuid4()).replace('-', '')
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run_request = {
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"header": {
|
||||
"message_id": run_message_id,
|
||||
"task_id": task_id,
|
||||
"namespace": "FlowingSpeechSynthesizer",
|
||||
"name": "RunSynthesis",
|
||||
"appkey": self.appkey,
|
||||
},
|
||||
"payload": {
|
||||
"text": text
|
||||
}
|
||||
}
|
||||
|
||||
await ws_connection.send(json.dumps(run_request))
|
||||
logger.bind(tag=TAG).debug(f"发送RunSynthesis指令: {text} (message_id: {run_message_id})")
|
||||
|
||||
# 立即发送StopSynthesis指令，避免IDLE_TIMEOUT
|
||||
stop_message_id = str(uuid.uuid4()).replace('-', '')
|
||||
stop_request = {
|
||||
"header": {
|
||||
"message_id": stop_message_id,
|
||||
"task_id": task_id,
|
||||
"namespace": "FlowingSpeechSynthesizer",
|
||||
"name": "StopSynthesis",
|
||||
"appkey": self.appkey,
|
||||
}
|
||||
}
|
||||
|
||||
await ws_connection.send(json.dumps(stop_request))
|
||||
logger.bind(tag=TAG).debug(f"发送StopSynthesis指令: {stop_message_id}")
|
||||
|
||||
# 初始化缓冲区和计数器
|
||||
self.pcm_buffer.clear()
|
||||
self.segment_count = 0 # 重置分段计数器
|
||||
opus_datas_cache = []
|
||||
|
||||
# 发送第一个音频包
|
||||
self.tts_audio_queue.put((SentenceType.FIRST, [], text))
|
||||
|
||||
# 标记是否需要发送StopSynthesis
|
||||
synthesis_completed = False
|
||||
|
||||
# 处理响应 - 设置超时避免长时间等待
|
||||
try:
|
||||
async def process_messages():
|
||||
nonlocal synthesis_completed
|
||||
async for message in ws_connection:
|
||||
try:
|
||||
if isinstance(message, str):
|
||||
# 处理JSON消息
|
||||
data = json.loads(message)
|
||||
header = data.get("header", {})
|
||||
event_name = header.get("name")
|
||||
|
||||
if event_name == "SynthesisStarted":
|
||||
logger.bind(tag=TAG).debug(f"TTS合成已启动: {task_id}")
|
||||
|
||||
elif event_name == "SynthesisCompleted":
|
||||
logger.bind(tag=TAG).debug(f"TTS合成完成: {text}")
|
||||
synthesis_completed = True
|
||||
break
|
||||
|
||||
elif event_name == "TaskFailed":
|
||||
error_msg = header.get("status_text", "未知错误")
|
||||
logger.bind(tag=TAG).error(f"TTS合成失败: {error_msg}")
|
||||
synthesis_completed = True
|
||||
break
|
||||
|
||||
elif isinstance(message, bytes):
|
||||
# 处理二进制音频数据
|
||||
self.pcm_buffer.extend(message)
|
||||
|
||||
# 计算每帧的字节数
|
||||
frame_bytes = int(
|
||||
self.opus_encoder.sample_rate
|
||||
* self.opus_encoder.channels
|
||||
* self.opus_encoder.frame_size_ms
|
||||
/ 1000
|
||||
* 2
|
||||
)
|
||||
|
||||
# 分帧处理PCM数据
|
||||
while len(self.pcm_buffer) >= frame_bytes:
|
||||
frame = bytes(self.pcm_buffer[:frame_bytes])
|
||||
del self.pcm_buffer[:frame_bytes]
|
||||
|
||||
# 编码为Opus
|
||||
opus_packets = self.opus_encoder.encode_pcm_to_opus(frame, False)
|
||||
if opus_packets:
|
||||
if self.segment_count < 10:
|
||||
self.tts_audio_queue.put(
|
||||
(SentenceType.MIDDLE, opus_packets, None)
|
||||
)
|
||||
self.segment_count += 1
|
||||
else:
|
||||
opus_datas_cache.extend(opus_packets)
|
||||
|
||||
except json.JSONDecodeError:
|
||||
logger.bind(tag=TAG).warning("收到无效的JSON消息")
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"处理响应消息失败: {e}")
|
||||
|
||||
await asyncio.wait_for(process_messages(), timeout=15) # 15秒超时
|
||||
|
||||
except asyncio.TimeoutError:
|
||||
logger.bind(tag=TAG).warning(f"TTS请求超时，但可能已获取部分音频数据: {text}")
|
||||
except websockets.ConnectionClosed:
|
||||
logger.bind(tag=TAG).debug("WebSocket连接已正常关闭")
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"处理WebSocket消息失败: {e}")
|
||||
|
||||
# 因为已经提前发送了StopSynthesis，这里不需要再次发送
|
||||
# 直接处理剩余的PCM数据
|
||||
if self.pcm_buffer:
|
||||
opus_packets = self.opus_encoder.encode_pcm_to_opus(
|
||||
bytes(self.pcm_buffer), end_of_stream=True
|
||||
)
|
||||
if opus_packets:
|
||||
if self.segment_count < 10:
|
||||
self.tts_audio_queue.put(
|
||||
(SentenceType.MIDDLE, opus_packets, None)
|
||||
)
|
||||
self.segment_count += 1
|
||||
else:
|
||||
opus_datas_cache.extend(opus_packets)
|
||||
self.pcm_buffer.clear()
|
||||
|
||||
# 发送缓存的数据
|
||||
if self.segment_count >= 10 and opus_datas_cache:
|
||||
self.tts_audio_queue.put(
|
||||
(SentenceType.MIDDLE, opus_datas_cache, None)
|
||||
)
|
||||
|
||||
# 如果是最后一段，处理待播放文件
|
||||
if is_last:
|
||||
self._process_before_stop_play_files()
|
||||
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"TTS异步请求异常: {e}")
|
||||
self.tts_audio_queue.put((SentenceType.LAST, [], None))
|
||||
|
||||
async def _tts_request_unified(self, text: str, is_last: bool) -> None:
|
||||
"""统一的TTS请求方法"""
|
||||
ws_connection = None
|
||||
|
||||
try:
|
||||
# 确保Token有效
|
||||
if self._is_token_expired():
|
||||
self._refresh_token()
|
||||
|
||||
# 总是创建独立连接，避免与其他线程的事件循环冲突
|
||||
ws_url = f"wss://{self.host}/ws/v1"
|
||||
ws_connection = await asyncio.wait_for(
|
||||
websockets.connect(
|
||||
ws_url,
|
||||
additional_headers={
|
||||
"X-NLS-Token": self.token,
|
||||
},
|
||||
ping_interval=15, # 每15秒发送ping，保持连接活跃
|
||||
ping_timeout=5, # ping超时时间5秒
|
||||
close_timeout=3, # 关闭超时时间3秒
|
||||
),
|
||||
timeout=8.0 # 连接超时时间8秒
|
||||
)
|
||||
logger.bind(tag=TAG).debug(f"建立独立WebSocket连接: {text}")
|
||||
|
||||
# 生成task_id
|
||||
task_id = str(uuid.uuid4()).replace('-', '')
|
||||
|
||||
# 第一阶段：发送StartSynthesis指令（设置参数）
|
||||
start_message_id = str(uuid.uuid4()).replace('-', '')
|
||||
start_request = {
|
||||
"header": {
|
||||
"message_id": start_message_id,
|
||||
"task_id": task_id,
|
||||
"namespace": "FlowingSpeechSynthesizer",
|
||||
"name": "StartSynthesis",
|
||||
"appkey": self.appkey,
|
||||
},
|
||||
"payload": {
|
||||
"voice": self.voice,
|
||||
"format": self.format,
|
||||
"sample_rate": self.sample_rate,
|
||||
"volume": self.volume,
|
||||
"speech_rate": self.speech_rate,
|
||||
"pitch_rate": self.pitch_rate,
|
||||
}
|
||||
}
|
||||
|
||||
await ws_connection.send(json.dumps(start_request))
|
||||
logger.bind(tag=TAG).debug(f"发送StartSynthesis指令: {start_message_id}")
|
||||
|
||||
# 第二阶段：发送RunSynthesis指令（发送文本）
|
||||
run_message_id = str(uuid.uuid4()).replace('-', '')
|
||||
run_request = {
|
||||
"header": {
|
||||
"message_id": run_message_id,
|
||||
"task_id": task_id,
|
||||
"namespace": "FlowingSpeechSynthesizer",
|
||||
"name": "RunSynthesis",
|
||||
"appkey": self.appkey,
|
||||
},
|
||||
"payload": {
|
||||
"text": text
|
||||
}
|
||||
}
|
||||
|
||||
await ws_connection.send(json.dumps(run_request))
|
||||
logger.bind(tag=TAG).debug(f"发送RunSynthesis指令: {text} (message_id: {run_message_id})")
|
||||
|
||||
# 立即发送StopSynthesis指令，避免IDLE_TIMEOUT
|
||||
stop_message_id = str(uuid.uuid4()).replace('-', '')
|
||||
stop_request = {
|
||||
"header": {
|
||||
"message_id": stop_message_id,
|
||||
"task_id": task_id,
|
||||
"namespace": "FlowingSpeechSynthesizer",
|
||||
"name": "StopSynthesis",
|
||||
"appkey": self.appkey,
|
||||
}
|
||||
}
|
||||
|
||||
await ws_connection.send(json.dumps(stop_request))
|
||||
logger.bind(tag=TAG).debug(f"发送StopSynthesis指令: {stop_message_id}")
|
||||
|
||||
# 初始化处理参数 - 使用独立缓冲区
|
||||
pcm_buffer = bytearray()
|
||||
opus_datas_cache = []
|
||||
segment_count = 0
|
||||
self.segment_count = 0 # 同时重置实例变量
|
||||
synthesis_completed = False
|
||||
|
||||
# 发送第一个音频包
|
||||
self.tts_audio_queue.put((SentenceType.FIRST, [], text))
|
||||
|
||||
# 处理响应 - 设置超时时间避免长时间等待
|
||||
timeout_duration = 15 # 15秒超时
|
||||
try:
|
||||
# 使用asyncio.wait_for替代asyncio.timeout以保证兼容性
|
||||
async def process_messages():
|
||||
nonlocal synthesis_completed, segment_count, pcm_buffer, opus_datas_cache # 声明使用外层变量
|
||||
async for message in ws_connection:
|
||||
try:
|
||||
if isinstance(message, str):
|
||||
# 处理JSON消息
|
||||
data = json.loads(message)
|
||||
header = data.get("header", {})
|
||||
event_name = header.get("name")
|
||||
|
||||
if event_name == "SynthesisStarted":
|
||||
logger.bind(tag=TAG).debug(f"TTS合成已启动: {task_id}")
|
||||
|
||||
elif event_name == "SynthesisCompleted":
|
||||
logger.bind(tag=TAG).debug(f"TTS合成完成: {text}")
|
||||
synthesis_completed = True
|
||||
break
|
||||
|
||||
elif event_name == "TaskFailed":
|
||||
error_msg = header.get("status_text", "未知错误")
|
||||
logger.bind(tag=TAG).error(f"TTS合成失败: {error_msg}")
|
||||
synthesis_completed = True
|
||||
break
|
||||
|
||||
elif isinstance(message, bytes):
|
||||
# 处理二进制音频数据
|
||||
pcm_buffer.extend(message)
|
||||
|
||||
# 计算每帧的字节数
|
||||
frame_bytes = int(
|
||||
self.opus_encoder.sample_rate
|
||||
* self.opus_encoder.channels
|
||||
* self.opus_encoder.frame_size_ms
|
||||
/ 1000
|
||||
* 2
|
||||
)
|
||||
|
||||
# 分帧处理PCM数据
|
||||
while len(pcm_buffer) >= frame_bytes:
|
||||
frame = bytes(pcm_buffer[:frame_bytes])
|
||||
del pcm_buffer[:frame_bytes] # 清除已处理的数据
|
||||
|
||||
# 编码为Opus
|
||||
opus_packets = self.opus_encoder.encode_pcm_to_opus(frame, False)
|
||||
if opus_packets:
|
||||
if segment_count < 10:
|
||||
self.tts_audio_queue.put(
|
||||
(SentenceType.MIDDLE, opus_packets, None)
|
||||
)
|
||||
segment_count += 1
|
||||
else:
|
||||
opus_datas_cache.extend(opus_packets)
|
||||
|
||||
except json.JSONDecodeError:
|
||||
logger.bind(tag=TAG).warning("收到无效的JSON消息")
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"处理响应消息失败: {e}")
|
||||
|
||||
await asyncio.wait_for(process_messages(), timeout=timeout_duration)
|
||||
|
||||
except asyncio.TimeoutError:
|
||||
logger.bind(tag=TAG).warning(f"TTS请求超时，但可能已获取部分音频数据: {text}")
|
||||
except websockets.ConnectionClosed:
|
||||
logger.bind(tag=TAG).debug("WebSocket连接已正常关闭")
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"处理WebSocket消息失败: {e}")
|
||||
|
||||
# 因为已经提前发送了StopSynthesis，这里不需要再次发送
|
||||
# 直接处理剩余的PCM数据
|
||||
if pcm_buffer:
|
||||
opus_packets = self.opus_encoder.encode_pcm_to_opus(
|
||||
bytes(pcm_buffer), end_of_stream=True
|
||||
)
|
||||
if opus_packets:
|
||||
if segment_count < 10:
|
||||
self.tts_audio_queue.put(
|
||||
(SentenceType.MIDDLE, opus_packets, None)
|
||||
)
|
||||
segment_count += 1
|
||||
else:
|
||||
opus_datas_cache.extend(opus_packets)
|
||||
|
||||
# 发送缓存的数据
|
||||
if segment_count >= 10 and opus_datas_cache:
|
||||
self.tts_audio_queue.put(
|
||||
(SentenceType.MIDDLE, opus_datas_cache, None)
|
||||
)
|
||||
|
||||
# 如果是最后一段，处理待播放文件
|
||||
if is_last:
|
||||
self._process_before_stop_play_files()
|
||||
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
|
||||
self.tts_audio_queue.put((SentenceType.LAST, [], None))
|
||||
finally:
|
||||
# 确保WebSocket连接被关闭
|
||||
if ws_connection:
|
||||
try:
|
||||
if hasattr(ws_connection, 'close'):
|
||||
if asyncio.iscoroutinefunction(ws_connection.close):
|
||||
await ws_connection.close()
|
||||
else:
|
||||
ws_connection.close()
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).debug(f"关闭WebSocket连接时出现异常: {e}")
|
||||
async def close(self):
|
||||
"""资源清理"""
|
||||
if self.ws:
|
||||
try:
|
||||
# 兼容不同版本的websockets库关闭方式
|
||||
if hasattr(self.ws, 'close'):
|
||||
if asyncio.iscoroutinefunction(self.ws.close):
|
||||
await self.ws.close()
|
||||
else:
|
||||
self.ws.close()
|
||||
elif hasattr(self.ws, 'close_connection'):
|
||||
await self.ws.close_connection()
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).debug(f"关闭WebSocket连接时出现异常: {e}")
|
||||
finally:
|
||||
self.ws = None
|
||||
|
||||
if hasattr(self, "opus_encoder"):
|
||||
self.opus_encoder.close()
|
||||
|
||||
await super().close()
|
||||
|
||||
def _process_before_stop_play_files(self):
|
||||
"""处理停止前的待播放文件"""
|
||||
for tts_file, text in self.before_stop_play_files:
|
||||
if tts_file and os.path.exists(tts_file):
|
||||
audio_datas = self._process_audio_file(tts_file)
|
||||
self.tts_audio_queue.put((SentenceType.MIDDLE, audio_datas, text))
|
||||
self.before_stop_play_files.clear()
|
||||
self.tts_audio_queue.put((SentenceType.LAST, [], None))
|
||||
|
||||
def _process_audio_file(self, tts_file):
|
||||
"""处理音频文件并转换为指定格式"""
|
||||
audio_datas = []
|
||||
if self.conn.audio_format == "pcm":
|
||||
audio_datas, _ = self.audio_to_pcm_data(tts_file)
|
||||
else:
|
||||
audio_datas, _ = self.audio_to_opus_data(tts_file)
|
||||
|
||||
if (
|
||||
self.delete_audio_file
|
||||
and tts_file is not None
|
||||
and os.path.exists(tts_file)
|
||||
and tts_file.startswith(self.output_file)
|
||||
):
|
||||
os.remove(tts_file)
|
||||
return audio_datas
|
||||
|
||||
def to_tts(self, text: str) -> list:
|
||||
"""非流式TTS处理，用于测试及保存音频文件的场景"""
|
||||
start_time = time.time()
|
||||
text = MarkdownCleaner.clean_markdown(text)
|
||||
|
||||
try:
|
||||
# 使用同步方式进行TTS转换
|
||||
if self._is_token_expired():
|
||||
self._refresh_token()
|
||||
|
||||
# 构造请求数据
|
||||
request_json = {
|
||||
"appkey": self.appkey,
|
||||
"token": self.token,
|
||||
"text": text,
|
||||
"format": "pcm",
|
||||
"sample_rate": self.sample_rate,
|
||||
"voice": self.voice,
|
||||
"volume": self.volume,
|
||||
"speech_rate": self.speech_rate,
|
||||
"pitch_rate": self.pitch_rate,
|
||||
}
|
||||
|
||||
# 使用HTTP接口进行同步请求
|
||||
import requests
|
||||
api_url = f"https://{self.host}/stream/v1/tts"
|
||||
headers = {"Content-Type": "application/json"}
|
||||
|
||||
resp = requests.post(api_url, json=request_json, headers=headers)
|
||||
|
||||
if resp.status_code == 401: # Token过期特殊处理
|
||||
self._refresh_token()
|
||||
resp = requests.post(api_url, json=request_json, headers=headers)
|
||||
|
||||
if resp.headers["Content-Type"].startswith("audio/"):
|
||||
pcm_data = resp.content
|
||||
|
||||
# 使用opus编码器处理PCM数据
|
||||
opus_datas = []
|
||||
frame_bytes = int(
|
||||
self.opus_encoder.sample_rate
|
||||
* self.opus_encoder.channels
|
||||
* self.opus_encoder.frame_size_ms
|
||||
/ 1000
|
||||
* 2
|
||||
)
|
||||
|
||||
# 分帧处理PCM数据
|
||||
for i in range(0, len(pcm_data), frame_bytes):
|
||||
frame = pcm_data[i:i + frame_bytes]
|
||||
if len(frame) == frame_bytes:
|
||||
opus = self.opus_encoder.encode_pcm_to_opus(frame, False)
|
||||
if opus:
|
||||
opus_datas.extend(opus)
|
||||
|
||||
logger.bind(tag=TAG).info(f"TTS请求成功: {text}, 耗时: {time.time() - start_time}秒")
|
||||
return opus_datas
|
||||
else:
|
||||
logger.bind(tag=TAG).error(f"TTS请求失败: {resp.content}")
|
||||
return []
|
||||
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
|
||||
return []
|
||||
|
||||
def _get_segment_text(self):
|
||||
"""获取当前可以处理的文本段"""
|
||||
if not self.tts_text_buff:
|
||||
return None
|
||||
|
||||
full_text = "".join(self.tts_text_buff)
|
||||
if len(full_text) <= self.processed_chars:
|
||||
return None
|
||||
|
||||
# 获取未处理的文本
|
||||
remaining_text = full_text[self.processed_chars:]
|
||||
|
||||
# 如果文本较短或者到达了句子结尾标点，直接处理
|
||||
sentence_endings = ['。', '！', '？', '.', '!', '?', '\n']
|
||||
if len(remaining_text) < 20:
|
||||
return None
|
||||
|
||||
# 查找句子结尾
|
||||
for i, char in enumerate(remaining_text):
|
||||
if char in sentence_endings and i > 10: # 至少10个字符
|
||||
segment = remaining_text[:i+1]
|
||||
segment_text = textUtils.get_string_no_punctuation_or_emoji(segment)
|
||||
if segment_text:
|
||||
self.processed_chars += i + 1
|
||||
return segment_text
|
||||
|
||||
# 如果没有找到句子结尾，但文本足够长，按长度分段
|
||||
if len(remaining_text) > 50:
|
||||
segment = remaining_text[:30]
|
||||
segment_text = textUtils.get_string_no_punctuation_or_emoji(segment)
|
||||
if segment_text:
|
||||
self.processed_chars += 30
|
||||
return segment_text
|
||||
|
||||
return None
|
||||
Reference in New Issue
Block a user