import os
|
import time
|
import uuid
|
import json
|
import hmac
|
import queue
|
import base64
|
import hashlib
|
import asyncio
|
import traceback
|
import websockets
|
|
from asyncio import Task
|
from typing import Callable, Any
|
from config.logger import setup_logging
|
from core.utils.tts import MarkdownCleaner
|
from urllib.parse import urlencode, urlparse
|
from core.providers.tts.base import TTSProviderBase
|
from core.providers.tts.dto.dto import SentenceType, ContentType, InterfaceType
|
|
TAG = __name__
|
logger = setup_logging()
|
|
|
class XunfeiWSAuth:
|
@staticmethod
|
def create_auth_url(api_key, api_secret, api_url):
|
"""生成讯飞WebSocket认证URL"""
|
parsed_url = urlparse(api_url)
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host = parsed_url.netloc
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path = parsed_url.path
|
|
# 获取UTC时间，讯飞要求使用RFC1123格式
|
now = time.gmtime()
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date = time.strftime('%a, %d %b %Y %H:%M:%S GMT', now)
|
|
# 构造签名字符串
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signature_origin = f"host: {host}\ndate: {date}\nGET {path} HTTP/1.1"
|
|
# 计算签名
|
signature_sha = hmac.new(
|
api_secret.encode('utf-8'),
|
signature_origin.encode('utf-8'),
|
digestmod=hashlib.sha256
|
).digest()
|
signature_sha_base64 = base64.b64encode(signature_sha).decode(encoding='utf-8')
|
|
# 构造authorization
|
authorization_origin = f'api_key="{api_key}", algorithm="hmac-sha256", headers="host date request-line", signature="{signature_sha_base64}"'
|
authorization = base64.b64encode(authorization_origin.encode('utf-8')).decode(encoding='utf-8')
|
|
# 构造最终的WebSocket URL
|
v = {
|
"authorization": authorization,
|
"date": date,
|
"host": host
|
}
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url = api_url + '?' + urlencode(v)
|
return url
|
|
|
class TTSProvider(TTSProviderBase):
|
TTS_PARAM_CONFIG = [
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("ttsVolume", "volume", 0, 100, 50, int),
|
("ttsRate", "speed", 0, 100, 50, int),
|
("ttsPitch", "pitch", 0, 100, 50, int),
|
]
|
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def __init__(self, config, delete_audio_file):
|
super().__init__(config, delete_audio_file)
|
|
# 设置为流式接口类型
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self.interface_type = InterfaceType.DUAL_STREAM
|
|
# 基础配置
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self.app_id = config.get("app_id")
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self.api_key = config.get("api_key")
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self.api_secret = config.get("api_secret")
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self.report_on_last = True
|
|
# 接口地址
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self.api_url = config.get("api_url", "wss://cbm01.cn-huabei-1.xf-yun.com/v1/private/mcd9m97e6")
|
|
# 音色配置
|
self.voice = config.get("voice", "x5_lingxiaoxuan_flow")
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if config.get("private_voice"):
|
self.voice = config.get("private_voice")
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# 音频参数配置
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speed = config.get("speed", "50")
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self.speed = int(speed) if speed else 50
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|
volume = config.get("volume", "50")
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self.volume = int(volume) if volume else 50
|
|
pitch = config.get("pitch", "50")
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self.pitch = int(pitch) if pitch else 50
|
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# 应用百分比调整（如果存在），否则使用公有化配置
|
self._apply_percentage_params(config)
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|
# 音频编码配置
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self.format = config.get("format", "raw")
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# 口语化配置
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self.oral_level = config.get("oral_level", "mid")
|
|
spark_assist = config.get("spark_assist", "1")
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self.spark_assist = int(spark_assist) if spark_assist else 1
|
|
stop_split = config.get("stop_split", "0")
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self.stop_split = int(stop_split) if stop_split else 0
|
|
remain = config.get("remain", "0")
|
self.remain = int(remain) if remain else 0
|
|
# WebSocket配置
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self.ws = None
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self._monitor_task = None
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self.activate_session = False
|
|
# 序列号管理
|
self.text_seq = 0
|
|
# 验证必需参数
|
if not all([self.app_id, self.api_key, self.api_secret]):
|
raise ValueError("讯飞TTS需要配置app_id、api_key和api_secret")
|
|
async def _ensure_connection(self):
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"""确保WebSocket连接可用"""
|
try:
|
logger.bind(tag=TAG).debug("开始建立新连接...")
|
|
# 生成认证URL
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auth_url = XunfeiWSAuth.create_auth_url(
|
self.api_key, self.api_secret, self.api_url
|
)
|
|
self.ws = await websockets.connect(
|
auth_url,
|
ping_interval=30,
|
ping_timeout=10,
|
close_timeout=10,
|
)
|
logger.bind(tag=TAG).debug("WebSocket连接建立成功")
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return self.ws
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except Exception as e:
|
logger.bind(tag=TAG).error(f"建立连接失败: {str(e)}")
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self.ws = None
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raise
|
|
def tts_text_priority_thread(self):
|
"""流式文本处理线程"""
|
while not self.conn.stop_event.is_set():
|
try:
|
message = self.tts_text_queue.get(timeout=1)
|
|
if self.conn.client_abort:
|
logger.bind(tag=TAG).info("收到打断信息，终止TTS文本处理线程")
|
continue
|
|
# 过滤旧消息：检查sentence_id是否匹配
|
if message.sentence_id != self.conn.sentence_id:
|
continue
|
|
logger.bind(tag=TAG).debug(
|
f"收到TTS任务｜{message.sentence_type.name} ｜ {message.content_type.name} | 会话ID: {message.sentence_id}"
|
)
|
|
if message.sentence_type == SentenceType.FIRST:
|
# 重置流式处理状态
|
self.reset_stream_state()
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# 重置序列号
|
self.text_seq = 0
|
# 增加序列号
|
self.text_seq += 1
|
|
if message.sentence_type == SentenceType.FIRST:
|
# 初始化参数
|
try:
|
if not getattr(self.conn, "sentence_id", None):
|
self.conn.sentence_id = uuid.uuid4().hex
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logger.bind(tag=TAG).debug(f"自动生成新的 会话ID: {self.conn.sentence_id}")
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|
logger.bind(tag=TAG).debug("开始启动TTS会话...")
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future = asyncio.run_coroutine_threadsafe(
|
self.start_session(self.conn.sentence_id),
|
loop=self.conn.loop,
|
)
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future.result(timeout=self.tts_timeout)
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self.before_stop_play_files.clear()
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logger.bind(tag=TAG).debug("TTS会话启动成功")
|
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"启动TTS会话失败: {str(e)}")
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continue
|
|
# 处理文本内容
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if ContentType.TEXT == message.content_type:
|
if message.content_detail:
|
try:
|
logger.bind(tag=TAG).debug(
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f"开始发送TTS文本: {message.content_detail}"
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)
|
future = asyncio.run_coroutine_threadsafe(
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self.text_to_speak(message.content_detail, None),
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loop=self.conn.loop,
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)
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future.result(timeout=self.tts_timeout)
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except Exception as e:
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logger.bind(tag=TAG).error(f"发送TTS文本失败: {str(e)}")
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# 不使用continue，确保后续处理不被中断
|
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# 处理文件内容
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if ContentType.FILE == message.content_type:
|
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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if message.content_file and os.path.exists(message.content_file):
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# 先处理文件音频数据
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self._process_audio_file_stream(message.content_file, callback=lambda audio_data: self.handle_audio_file(audio_data, message.content_detail))
|
|
# 处理会话结束
|
if message.sentence_type == SentenceType.LAST:
|
try:
|
logger.bind(tag=TAG).debug("开始结束TTS会话...")
|
asyncio.run_coroutine_threadsafe(
|
self.finish_session(self.conn.sentence_id),
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loop=self.conn.loop,
|
)
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"结束TTS会话失败: {str(e)}")
|
continue
|
|
except queue.Empty:
|
continue
|
except Exception as e:
|
logger.bind(tag=TAG).error(
|
f"处理TTS文本失败: {str(e)}, 类型: {type(e).__name__}, 堆栈: {traceback.format_exc()}"
|
)
|
|
async def text_to_speak(self, text, _):
|
"""发送文本到TTS服务进行合成"""
|
try:
|
if self.ws is None:
|
logger.bind(tag=TAG).warning(f"WebSocket连接不存在，终止发送文本")
|
return
|
|
filtered_text = MarkdownCleaner.clean_markdown(text)
|
|
if filtered_text:
|
# 使用滑动窗口匹配处理跨分片的替换词
|
confirmed_texts, self._pending_prefix = self._match_stream_text(filtered_text)
|
|
# 发送每个确定的文本片段
|
for txt in confirmed_texts:
|
if txt and self.ws:
|
# 发送文本合成请求
|
run_request = self._build_base_request(status=1, text=txt)
|
await self.ws.send(json.dumps(run_request))
|
return
|
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"发送TTS文本失败: {str(e)}")
|
if self.ws:
|
try:
|
await self.ws.close()
|
except:
|
pass
|
self.ws = None
|
raise
|
|
async def start_session(self, session_id):
|
logger.bind(tag=TAG).debug(f"开始会话～～{session_id}")
|
try:
|
# 上个会话处于激活状态时关闭上个连接新建链接
|
if self.activate_session:
|
await self.close()
|
|
# 设置会话激活标志
|
self.activate_session = True
|
|
# 建立新连接
|
await self._ensure_connection()
|
|
# 启动监听任务
|
if self._monitor_task is None or self._monitor_task.done():
|
logger.bind(tag=TAG).debug("启动监听任务...")
|
self._monitor_task = asyncio.create_task(self._start_monitor_tts_response())
|
|
# 发送会话启动请求
|
start_request = self._build_base_request(status=0)
|
|
await self.ws.send(json.dumps(start_request))
|
logger.bind(tag=TAG).debug("会话启动请求已发送")
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"启动会话失败: {str(e)}")
|
# 确保清理资源
|
await self.close()
|
raise
|
|
async def finish_session(self, session_id):
|
logger.bind(tag=TAG).debug(f"关闭会话～～{session_id}")
|
try:
|
if self.ws:
|
# 发送会话结束请求
|
stop_request = self._build_base_request(status=2)
|
await self.ws.send(json.dumps(stop_request))
|
logger.bind(tag=TAG).debug("会话结束请求已发送")
|
|
if self._monitor_task:
|
try:
|
await self._monitor_task
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"等待监听任务完成时发生错误: {str(e)}")
|
finally:
|
self._monitor_task = None
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"关闭会话失败: {str(e)}")
|
await self.close()
|
raise
|
|
async def close(self):
|
"""资源清理"""
|
await super().close()
|
self.activate_session = False
|
if self._monitor_task:
|
try:
|
self._monitor_task.cancel()
|
await self._monitor_task
|
except asyncio.CancelledError:
|
pass
|
except Exception as e:
|
logger.bind(tag=TAG).warning(f"关闭时取消监听任务错误: {e}")
|
self._monitor_task = None
|
|
if self.ws:
|
try:
|
await self.ws.close()
|
except:
|
pass
|
self.ws = None
|
|
async def _start_monitor_tts_response(self):
|
"""监听TTS响应"""
|
try:
|
while not self.conn.stop_event.is_set():
|
try:
|
msg = await self.ws.recv()
|
|
# 检查客户端是否中止
|
if self.conn.client_abort:
|
logger.bind(tag=TAG).info("收到打断信息，终止监听TTS响应")
|
break
|
|
try:
|
data = json.loads(msg)
|
header = data.get("header", {})
|
code = header.get("code")
|
|
if code == 0:
|
payload = data.get("payload", {})
|
audio_payload = payload.get("audio", {})
|
|
if audio_payload:
|
status = audio_payload.get("status", 0)
|
audio_data = audio_payload.get("audio", "")
|
if status == 0:
|
logger.bind(tag=TAG).debug("TTS合成已启动")
|
self.tts_audio_queue.put(
|
(SentenceType.FIRST, [], None)
|
)
|
elif status == 2:
|
logger.bind(tag=TAG).debug("收到结束状态的音频数据，TTS合成完成")
|
self.activate_session = False
|
self._process_before_stop_play_files()
|
break
|
else:
|
tts_text = self.get_tts_text(self.conn.sentence_id)
|
if tts_text:
|
logger.bind(tag=TAG).info(
|
f"句子语音生成成功： {tts_text}"
|
)
|
self.tts_audio_queue.put(
|
(SentenceType.FIRST, [], tts_text)
|
)
|
self.clear_tts_text(self.conn.sentence_id)
|
try:
|
audio_bytes = base64.b64decode(audio_data)
|
self.opus_encoder.encode_pcm_to_opus_stream(
|
audio_bytes, False, self.handle_opus
|
)
|
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"处理音频数据失败: {e}")
|
|
else:
|
message = header.get("message", "未知错误")
|
logger.bind(tag=TAG).error(f"TTS合成错误: {code} - {message}")
|
break
|
|
except json.JSONDecodeError:
|
logger.bind(tag=TAG).warning("收到无效的JSON消息")
|
|
except websockets.ConnectionClosed:
|
logger.bind(tag=TAG).warning("WebSocket连接已关闭")
|
break
|
|
except Exception as e:
|
logger.bind(tag=TAG).error(
|
f"处理TTS响应时出错: {e}\n{traceback.format_exc()}"
|
)
|
break
|
|
# 链接不可复用
|
if self.ws:
|
try:
|
await self.ws.close()
|
except:
|
pass
|
self.ws = None
|
# 监听任务退出时清理引用
|
finally:
|
self.activate_session = False
|
self._monitor_task = None
|
|
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"""
|
try:
|
# 创建新的事件循环
|
loop = asyncio.new_event_loop()
|
asyncio.set_event_loop(loop)
|
|
# 存储音频数据
|
audio_data = []
|
|
async def _generate_audio():
|
# 生成认证URL
|
auth_url = XunfeiWSAuth.create_auth_url(
|
self.api_key, self.api_secret, self.api_url
|
)
|
|
# 建立WebSocket连接
|
ws = await websockets.connect(
|
auth_url,
|
ping_interval=30,
|
ping_timeout=10,
|
close_timeout=10,
|
)
|
|
try:
|
filtered_text = MarkdownCleaner.clean_markdown(text)
|
if self._correct_words_pattern:
|
filtered_text = self._correct_words_pattern.sub(lambda m: self.correct_words[m.group(0)], filtered_text)
|
|
text_request = self._build_base_request(status=2,text=filtered_text)
|
|
await ws.send(json.dumps(text_request))
|
|
task_finished = False
|
while not task_finished:
|
msg = await ws.recv()
|
|
data = json.loads(msg)
|
header = data.get("header", {})
|
code = header.get("code")
|
|
if code == 0:
|
payload = data.get("payload", {})
|
audio_payload = payload.get("audio", {})
|
if audio_payload:
|
status = audio_payload.get("status", 0)
|
audio_base64 = audio_payload.get("audio", "")
|
if status == 1:
|
try:
|
audio_bytes = base64.b64decode(audio_base64)
|
self.opus_encoder.encode_pcm_to_opus_stream(
|
audio_bytes,
|
end_of_stream=False,
|
callback=lambda opus: audio_data.append(opus)
|
)
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"处理音频数据失败: {e}")
|
elif status == 2:
|
task_finished = True
|
logger.bind(tag=TAG).debug("TTS任务完成")
|
|
else:
|
message = header.get("message", "未知错误")
|
raise Exception(f"合成失败: {code} - {message}")
|
|
finally:
|
# 清理资源
|
try:
|
await ws.close()
|
except:
|
pass
|
|
loop.run_until_complete(_generate_audio())
|
loop.close()
|
|
return audio_data
|
except Exception as e:
|
logger.bind(tag=TAG).error(f"生成音频数据失败: {str(e)}")
|
return []
|
|
def audio_to_opus_data_stream(
|
self, audio_file_path, callback: Callable[[Any], Any] = None
|
):
|
"""重写父类方法：使用独立的临时编码器处理音频文件，避免与TTS流式编码器并发冲突。
|
双流式TTS中，monitor任务在event loop线程接收TTS音频并使用self.opus_encoder编码，
|
同时tts_text_priority_thread处理音乐文件也使用self.opus_encoder，
|
共享的encoder.buffer非线程安全，并发访问会导致SILK resampler断言失败。
|
"""
|
from core.utils.util import audio_to_data_stream
|
|
return audio_to_data_stream(
|
audio_file_path,
|
is_opus=True,
|
callback=callback,
|
sample_rate=self.conn.sample_rate,
|
opus_encoder=None,
|
)
|
|
def _build_base_request(self, status, text=" "):
|
"""构建基础请求结构"""
|
return {
|
"header": {
|
"app_id": self.app_id,
|
"status": status,
|
},
|
"parameter": {
|
"oral": {
|
"oral_level": self.oral_level,
|
"spark_assist": self.spark_assist,
|
"stop_split": self.stop_split,
|
"remain": self.remain
|
},
|
"tts": {
|
"vcn": self.voice,
|
"speed": self.speed,
|
"volume": self.volume,
|
"pitch": self.pitch,
|
"bgs": 0,
|
"reg": 0,
|
"rdn": 0,
|
"rhy": 0,
|
"audio": {
|
"encoding": self.format,
|
"sample_rate": self.conn.sample_rate,
|
"channels": 1,
|
"bit_depth": 16,
|
"frame_size": 0
|
}
|
}
|
},
|
"payload": {
|
"text": {
|
"encoding": "utf8",
|
"compress": "raw",
|
"format": "plain",
|
"status": status,
|
"seq": self.text_seq,
|
"text": base64.b64encode(text.encode('utf-8')).decode('utf-8')
|
}
|
}
|
}
|
