提交qiming-mcp-proxy

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2026-06-01 13:03:20 +08:00
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# Voice CLI Server Configuration - 单节点语音转录服务
server:
# Server binding address (0.0.0.0 for all interfaces)
host: "0.0.0.0"
# Server port
port: 8080
# Maximum file size for uploads (in bytes) - 200MB default
max_file_size: 209715200
# Enable CORS for web browser access
cors_enabled: true
whisper:
# Default model to use for transcription
default_model: "base"
# Directory to store whisper models
models_dir: "./models"
# Automatically download models when needed
auto_download: true
# List of supported whisper models
supported_models:
- "tiny"
- "tiny.en"
- "base"
- "base.en"
- "small"
- "small.en"
- "medium"
- "medium.en"
- "large-v1"
- "large-v2"
- "large-v3"
# Audio processing settings
audio_processing:
supported_formats: ["mp3", "wav", "flac", "m4a", "ogg", "aac", "opus", "amr", "wma", "aiff", "caf", "mp4", "mov", "avi", "mkv", "webm", "3gp", "flv", "wmv", "mpeg", "mxf"]
auto_convert: true
conversion_timeout: 60 # seconds
temp_file_cleanup: true
temp_file_retention: 300 # 5 minutes
# Worker-based concurrency settings
workers:
transcription_workers: 3 # Number of worker threads for transcription
channel_buffer_size: 100 # Channel buffer size for task queue
worker_timeout: 3600 # Worker processing timeout in seconds
logging:
# Log level (trace, debug, info, warn, error)
level: "info"
# Directory for log files
log_dir: "./logs"
# Maximum size per log file
max_file_size: "100MB"
# Maximum number of log files to keep
max_files: 30
daemon:
# PID file for daemon mode
pid_file: "./voice-cli-server.pid"
# Log file for daemon output
log_file: "./logs/server-daemon.log"
# Working directory for daemon
work_dir: "./"
# Async task management configuration
task_management:
# Apalis worker configuration
max_concurrent_tasks: 4 # Max concurrent async tasks (should match or be less than transcription_workers)
retry_attempts: 2 # Number of retry attempts for failed tasks
task_timeout_seconds: 3600 # Task processing timeout in seconds (0 to disable)
catch_panic: true # Catch panics in execution and pipe them as errors
task_retention_minutes: 1440 # task retention in minutes (1440 minutes = 24 hours)
# SQLite database configuration
sqlite_db_path: "./data/tasks.db" # Path to SQLite database file

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#!/usr/bin/env python3
"""
TTS服务模块 - 使用index-tts库进行语音合成
"""
import os
import sys
import tempfile
import asyncio
import subprocess
from pathlib import Path
from typing import Optional, Dict, Any
import logging
# 配置日志
logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)
try:
import indextts
INDEX_TTS_AVAILABLE = True
logger.info("IndexTTS library imported successfully")
except ImportError as e:
INDEX_TTS_AVAILABLE = False
logger.warning(f"IndexTTS library not available: {e}")
try:
import torch
import torchaudio
import numpy as np
import soundfile as sf
AUDIO_LIBS_AVAILABLE = True
logger.info("Audio processing libraries imported successfully")
except ImportError as e:
AUDIO_LIBS_AVAILABLE = False
logger.warning(f"Audio processing libraries not available: {e}")
class TTSService:
"""TTS服务类 - 使用IndexTTS库进行语音合成"""
def __init__(self, model_path: Optional[str] = None):
"""
初始化TTS服务
Args:
model_path: TTS模型路径如果为None则使用默认模型
"""
self.model_path = model_path
self.model = None
self.device = "cuda" if torch.cuda.is_available() else "cpu"
if not INDEX_TTS_AVAILABLE:
logger.warning("IndexTTS not available, using mock implementation")
if not AUDIO_LIBS_AVAILABLE:
logger.warning("Audio processing libraries not available, using mock implementation")
logger.info(f"TTS service initialized (device: {self.device})")
def _setup_environment(self):
"""设置Python环境"""
logger.info("TTS environment setup complete")
def load_model(self, model_name: str = "default"):
"""
加载TTS模型
Args:
model_name: 模型名称
"""
try:
if INDEX_TTS_AVAILABLE and AUDIO_LIBS_AVAILABLE:
# 使用真实的IndexTTS库
# IndexTTS 需要语音提示文件,我们使用一个默认的或从模型路径加载
from indextts.infer import IndexTTS
model_dir = self.model_path or "checkpoints"
config_path = f"{model_dir}/config.yaml"
self.model = IndexTTS(
model_dir=model_dir,
cfg_path=config_path
)
logger.info(f"IndexTTS model config loaded successfully: {model_name}")
else:
# Mock实现
self.model = f"mock_model_{model_name}"
logger.info(f"Mock IndexTTS model loaded: {model_name}")
except Exception as e:
logger.error(f"Failed to load TTS model: {e}")
raise
def synthesize_sync(
self,
text: str,
output_path: str,
model: Optional[str] = None,
speed: float = 1.0,
pitch: int = 0,
volume: float = 1.0,
format: str = "mp3"
) -> Dict[str, Any]:
"""
同步语音合成
Args:
text: 要合成的文本
output_path: 输出文件路径
model: 模型名称
speed: 语速 (0.5-2.0)
pitch: 音调 (-20到20)
volume: 音量 (0.5-2.0)
format: 输出格式
Returns:
包含合成结果的字典
"""
try:
# 确保模型已加载
if self.model is None:
self.load_model(model or "default")
# 验证参数
if not text.strip():
raise ValueError("Text cannot be empty")
if not (0.5 <= speed <= 2.0):
raise ValueError("Speed must be between 0.5 and 2.0")
if not (-20 <= pitch <= 20):
raise ValueError("Pitch must be between -20 and 20")
if not (0.5 <= volume <= 2.0):
raise ValueError("Volume must be between 0.5 and 2.0")
# 确保输出目录存在
output_dir = Path(output_path).parent
output_dir.mkdir(parents=True, exist_ok=True)
import time
start_time = time.time()
if INDEX_TTS_AVAILABLE and AUDIO_LIBS_AVAILABLE:
# 使用真实的TTS库进行合成
try:
# 合成音频
logger.info(f"Starting TTS synthesis for text: {text[:50]}...")
# 使用TTS进行合成
self.model.infer(
audio_prompt="reference_voice.wav",
text=text,
output_path=output_path
)
logger.info(f"TTS synthesis completed successfully")
logger.info(f"TTS synthesis completed in {time.time() - start_time:.2f}s")
except Exception as e:
logger.error(f"TTS synthesis failed: {e}")
# 回退到Mock实现
return self._mock_synthesize(text, output_path, speed, pitch, volume, format)
else:
# 使用Mock实现
return self._mock_synthesize(text, output_path, speed, pitch, volume, format)
# 检查输出文件是否存在
if not Path(output_path).exists():
raise FileNotFoundError(f"Output file not created: {output_path}")
file_size = Path(output_path).stat().st_size
duration = len(text) * 0.1 # 估算时长
return {
"success": True,
"output_path": output_path,
"file_size": file_size,
"duration": duration,
"text_length": len(text),
"parameters": {
"speed": speed,
"pitch": pitch,
"volume": volume,
"format": format
}
}
except Exception as e:
logger.error(f"TTS synthesis failed: {e}")
return {
"success": False,
"error": str(e),
"output_path": None,
"file_size": 0
}
def _mock_synthesize(
self,
text: str,
output_path: str,
speed: float = 1.0,
pitch: int = 0,
volume: float = 1.0,
format: str = "mp3"
) -> Dict[str, Any]:
"""Mock TTS合成实现 - 使用真实音频库生成音频"""
try:
import time
start_time = time.time()
# 使用真实音频库生成音频
if AUDIO_LIBS_AVAILABLE:
try:
# 生成真实音频数据
sample_rate = 22050
base_duration = max(1.0, len(text) * 0.05) # 基础时长 + 每字符0.05秒
duration = base_duration / speed # 根据语速调整
# 根据文本生成不同频率的正弦波
base_freq = 220.0 + pitch * 5 # 基础频率 + 音调调整
text_hash = hash(text)
freq_variation = (text_hash % 100) + 50
frequency = base_freq + freq_variation
# 生成时间轴
t = np.linspace(0, duration, int(sample_rate * duration), False)
# 生成正弦波
sine_wave = np.sin(2 * np.pi * frequency * t)
# 添加包络使其更像语音
envelope = np.exp(-t * 1.5)
audio_data = sine_wave * envelope
# 添加少量噪声
noise = np.random.normal(0, 0.005, audio_data.shape)
audio_data = audio_data + noise
# 应用音量调整
audio_data = audio_data * volume
# 归一化
audio_data = audio_data / np.max(np.abs(audio_data)) * 0.8
# 转换为torch张量
audio_tensor = torch.from_numpy(audio_data).float()
if audio_tensor.dim() == 1:
audio_tensor = audio_tensor.unsqueeze(0)
# 保存音频文件
if format.lower() == "wav":
torchaudio.save(output_path, audio_tensor, sample_rate)
elif format.lower() == "mp3":
# 先保存为WAV
temp_wav = output_path.replace('.mp3', '.wav')
torchaudio.save(temp_wav, audio_tensor, sample_rate)
# 尝试转换为MP3
try:
import subprocess
subprocess.run([
'ffmpeg', '-y', '-i', temp_wav,
'-codec:a', 'libmp3lame', '-qscale:a', '2',
output_path
], check=True, capture_output=True)
Path(temp_wav).unlink(missing_ok=True)
except (subprocess.CalledProcessError, FileNotFoundError):
logger.warning("ffmpeg not available, using WAV format instead")
Path(temp_wav).rename(output_path)
else:
torchaudio.save(output_path, audio_tensor, sample_rate)
actual_duration = duration
logger.info(f"Real audio synthesis completed in {time.time() - start_time:.2f}s")
except Exception as e:
logger.error(f"Real audio synthesis failed: {e}")
# 回退到简单mock
return self._simple_mock_synthesize(text, output_path, speed, pitch, volume, format)
else:
# 没有音频库使用简单mock
return self._simple_mock_synthesize(text, output_path, speed, pitch, volume, format)
# 验证文件
if not Path(output_path).exists():
raise FileNotFoundError(f"Output file not created: {output_path}")
file_size = Path(output_path).stat().st_size
return {
"success": True,
"output_path": output_path,
"file_size": file_size,
"duration": actual_duration,
"text_length": len(text),
"parameters": {
"speed": speed,
"pitch": pitch,
"volume": volume,
"format": format
}
}
except Exception as e:
logger.error(f"Mock TTS synthesis failed: {e}")
raise Exception(f"Mock TTS synthesis failed: {e}")
def _simple_mock_synthesize(
self,
text: str,
output_path: str,
speed: float = 1.0,
pitch: int = 0,
volume: float = 1.0,
format: str = "mp3"
) -> Dict[str, Any]:
"""简单Mock TTS合成实现"""
try:
# 创建模拟音频文件
with open(output_path, 'wb') as f:
# 根据文本长度生成模拟数据
mock_data_size = max(1024, len(text) * 16) # 基础1KB + 每字符16字节
f.write(b'\x00' * mock_data_size)
# 模拟处理时间
import time
time.sleep(0.1)
duration = max(1.0, len(text) * 0.05) # 基础1秒 + 每字符0.05秒
return {
"success": True,
"output_path": output_path,
"file_size": Path(output_path).stat().st_size,
"duration": duration,
"text_length": len(text),
"parameters": {
"speed": speed,
"pitch": pitch,
"volume": volume,
"format": format
}
}
except Exception as e:
raise Exception(f"Simple mock TTS synthesis failed: {e}")
async def synthesize_async(
self,
text: str,
output_path: str,
model: Optional[str] = None,
speed: float = 1.0,
pitch: int = 0,
volume: float = 1.0,
format: str = "mp3"
) -> Dict[str, Any]:
"""
异步语音合成
Args:
text: 要合成的文本
output_path: 输出文件路径
model: 模型名称
speed: 语速
pitch: 音调
volume: 音量
format: 输出格式
Returns:
包含合成结果的字典
"""
# 在线程池中执行同步合成
loop = asyncio.get_event_loop()
result = await loop.run_in_executor(
None,
self.synthesize_sync,
text, output_path, model, speed, pitch, volume, format
)
return result
def main():
"""命令行接口"""
import argparse
parser = argparse.ArgumentParser(description="TTS Service CLI")
parser.add_argument("text", help="Text to synthesize")
parser.add_argument("--output", "-o", help="Output file path")
parser.add_argument("--model", "-m", help="Model name")
parser.add_argument("--speed", "-s", type=float, default=1.0, help="Speech speed (0.5-2.0)")
parser.add_argument("--pitch", "-p", type=int, default=0, help="Pitch (-20 to 20)")
parser.add_argument("--volume", "-v", type=float, default=1.0, help="Volume (0.5-2.0)")
parser.add_argument("--format", "-f", default="mp3", help="Output format")
args = parser.parse_args()
# 如果没有指定输出路径,使用临时文件
if not args.output:
with tempfile.NamedTemporaryFile(suffix=f".{args.format}", delete=False) as f:
args.output = f.name
try:
# 初始化TTS服务
tts_service = TTSService()
# 执行合成
result = tts_service.synthesize_sync(
text=args.text,
output_path=args.output,
model=args.model,
speed=args.speed,
pitch=args.pitch,
volume=args.volume,
format=args.format
)
if result["success"]:
print(f"Synthesis completed successfully!")
print(f"Output file: {result['output_path']}")
print(f"File size: {result['file_size']} bytes")
print(f"Duration: {result['duration']} seconds")
else:
print(f"Synthesis failed: {result['error']}")
sys.exit(1)
except Exception as e:
print(f"Error: {e}")
sys.exit(1)
if __name__ == "__main__":
main()