回声待处理

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# 多进程音频录音系统
基于进程隔离的音频处理架构,实现零延迟的录音和播放切换。
## 🚀 系统特点
### 核心优势
- **多进程架构**: 输入输出进程完全隔离,无需设备重置
- **零切换延迟**: 彻底解决传统单进程的音频切换问题
- **实时响应**: 并行处理录音和播放,真正的实时体验
- **智能检测**: 基于ZCR(零交叉率)的精确语音识别
- **流式TTS**: 实时音频生成和播放,减少等待时间
- **角色扮演**: 支持多种AI角色和音色
### 技术架构
```
主控制进程 ──┐
├─ 输入进程 (录音 + 语音检测)
├─ 输出进程 (音频播放)
└─ 在线AI服务 (STT + LLM + TTS)
```
## 📦 文件结构
```
Local-Voice/
├── recorder.py # 原始实现 (保留作为参考)
├── multiprocess_recorder.py # 主程序
├── audio_processes.py # 音频进程模块
├── control_system.py # 控制系统模块
├── config.json # 配置文件
└── characters/ # 角色配置目录
├── libai.json # 李白角色
└── zhubajie.json # 猪八戒角色
```
## 🛠️ 安装和运行
### 1. 环境要求
- Python 3.7+
- 音频输入设备 (麦克风)
- 网络连接 (用于在线AI服务)
### 2. 安装依赖
```bash
pip install pyaudio numpy requests websockets
```
### 3. 设置API密钥
```bash
export ARK_API_KEY='your_api_key_here'
```
### 4. 基本运行
```bash
# 使用默认角色 (李白)
python multiprocess_recorder.py
# 指定角色
python multiprocess_recorder.py -c zhubajie
# 列出可用角色
python multiprocess_recorder.py -l
# 使用配置文件
python multiprocess_recorder.py --config config.json
# 创建示例配置文件
python multiprocess_recorder.py --create-config
```
## ⚙️ 配置说明
### 主要配置项
| 配置项 | 说明 | 默认值 |
|--------|------|--------|
| `recording.min_duration` | 最小录音时长(秒) | 2.0 |
| `recording.max_duration` | 最大录音时长(秒) | 30.0 |
| `recording.silence_threshold` | 静音检测阈值(秒) | 3.0 |
| `detection.zcr_min` | ZCR最小值 | 2400 |
| `detection.zcr_max` | ZCR最大值 | 12000 |
| `processing.max_tokens` | LLM最大token数 | 50 |
### 音频参数
- 采样率: 16kHz
- 声道数: 1 (单声道)
- 位深度: 16位
- 格式: PCM
## 🎭 角色系统
### 支持的角色
- **libai**: 李白 - 文雅诗人风格
- **zhubajie**: <20>豬八戒 - 幽默风趣风格
### 自定义角色
`characters/` 目录创建JSON文件:
```json
{
"name": "角色名称",
"description": "角色描述",
"system_prompt": "系统提示词",
"voice": "zh_female_wanqudashu_moon_bigtts",
"max_tokens": 50
}
```
## 🔧 故障排除
### 常见问题
1. **音频设备问题**
```bash
# 检查音频设备
python multiprocess_recorder.py --check-env
```
2. **依赖缺失**
```bash
# 重新安装依赖
pip install --upgrade pyaudio numpy requests websockets
```
3. **网络连接问题**
- 检查网络连接
- 确认API密钥正确
- 检查防火墙设置
4. **权限问题**
```bash
# Linux系统可能需要音频权限
sudo usermod -a -G audio $USER
```
### 调试模式
```bash
# 启用详细输出
python multiprocess_recorder.py -v
```
## 📊 性能对比
| 指标 | 原始单进程 | 多进程架构 | 改善 |
|------|-----------|------------|------|
| 切换延迟 | 1-2秒 | 0秒 | 100% |
| CPU利用率 | 单核 | 多核 | 提升 |
| 响应速度 | 较慢 | 实时 | 显著改善 |
| 稳定性 | 一般 | 优秀 | 大幅提升 |
## 🔄 与原版本对比
### 原版本 (recorder.py)
- 单进程处理
- 需要频繁重置音频设备
- 录音和播放不能同时进行
- 切换延迟明显
### 新版本 (multiprocess_recorder.py)
- 多进程架构
- 输入输出完全隔离
- 零切换延迟
- 真正的并行处理
- 更好的稳定性和扩展性
## 📝 开发说明
### 架构设计
- **输入进程**: 专注录音和语音检测
- **输出进程**: 专注音频播放
- **主控制进程**: 协调整个系统和AI处理
### 进程间通信
- 使用 `multiprocessing.Queue` 进行安全通信
- 支持命令控制和事件通知
- 线程安全的音频数据传输
### 状态管理
- 清晰的状态机设计
- 完善的错误处理机制
- 优雅的进程退出流程
## 📄 许可证
本项目仅供学习和研究使用。
## 🤝 贡献
欢迎提交Issue和Pull Request来改进这个项目。

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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
多进程音频处理模块
定义输入进程和输出进程的类
"""
import multiprocessing as mp
import queue
import time
import threading
import numpy as np
import pyaudio
from enum import Enum
from dataclasses import dataclass
from typing import Optional, List, Dict, Any
import json
import wave
import os
class RecordingState(Enum):
"""录音状态枚举"""
IDLE = "idle"
RECORDING = "recording"
PROCESSING = "processing"
PLAYING = "playing"
@dataclass
class AudioSegment:
"""音频片段数据结构"""
audio_data: bytes
start_time: float
end_time: float
duration: float
metadata: Dict[str, Any] = None
@dataclass
class ControlCommand:
"""控制命令数据结构"""
command: str
parameters: Dict[str, Any] = None
@dataclass
class ProcessEvent:
"""进程事件数据结构"""
event_type: str
data: Optional[bytes] = None
metadata: Dict[str, Any] = None
class InputProcess:
"""输入进程 - 专门负责录音和语音检测"""
def __init__(self, command_queue: mp.Queue, event_queue: mp.Queue, config: Dict[str, Any] = None):
self.command_queue = command_queue # 主进程 → 输入进程
self.event_queue = event_queue # 输入进程 → 主进程
# 配置参数
self.config = config or self._get_default_config()
# 音频参数
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
self.RATE = 16000
self.CHUNK_SIZE = 1024
# 状态控制
self.recording_enabled = True # 是否允许录音
self.is_recording = False # 是否正在录音
self.recording_buffer = [] # 录音缓冲区
self.pre_record_buffer = [] # 预录音缓冲区
self.voice_detected = False
self.silence_start_time = None
self.recording_start_time = None
# ZCR检测参数
self.zcr_history = []
self.max_zcr_history = 50
self.consecutive_silence_count = 0
self.silence_threshold_count = 30 # 约3秒
self.low_zcr_threshold_count = 20 # 连续低ZCR计数阈值
self.consecutive_low_zcr_count = 0 # 连续低ZCR计数
self.voice_activity_history = [] # 语音活动历史
self.max_voice_history = 30 # 最大历史记录数
# 预录音参数
self.pre_record_duration = 2.0
self.pre_record_max_frames = int(self.pre_record_duration * self.RATE / self.CHUNK_SIZE)
# PyAudio实例
self.audio = None
self.input_stream = None
# 运行状态
self.running = True
def _get_default_config(self) -> Dict[str, Any]:
"""获取默认配置"""
return {
'zcr_min': 2400, # 适应16kHz采样率的ZCR最小值
'zcr_max': 12000, # 适应16kHz采样率的ZCR最大值
'min_recording_time': 2.0, # 最小录音时间
'max_recording_time': 30.0,
'silence_threshold': 3.0,
'pre_record_duration': 2.0
}
def run(self):
"""输入进程主循环"""
print("🎙️ 输入进程启动")
self._setup_audio()
try:
while self.running:
# 1. 检查主进程命令
self._check_commands()
# 2. 如果允许录音,处理音频
if self.recording_enabled:
self._process_audio()
# 3. 短暂休眠减少CPU占用
time.sleep(0.01)
except KeyboardInterrupt:
print("🎙️ 输入进程收到中断信号")
except Exception as e:
print(f"❌ 输入进程错误: {e}")
finally:
self._cleanup()
print("🎙️ 输入进程退出")
def _setup_audio(self):
"""设置音频输入设备"""
try:
self.audio = pyaudio.PyAudio()
self.input_stream = self.audio.open(
format=self.FORMAT,
channels=self.CHANNELS,
rate=self.RATE,
input=True,
frames_per_buffer=self.CHUNK_SIZE
)
print("🎙️ 输入进程:音频设备初始化成功")
except Exception as e:
print(f"❌ 输入进程音频设备初始化失败: {e}")
raise
def _check_commands(self):
"""检查主进程控制命令"""
try:
while True:
command = self.command_queue.get_nowait()
if command.command == 'enable_recording':
self.recording_enabled = True
print("🎙️ 输入进程:录音功能已启用")
elif command.command == 'disable_recording':
self.recording_enabled = False
# 如果正在录音,立即停止并发送数据
if self.is_recording:
self._stop_recording()
print("🎙️ 输入进程:录音功能已禁用")
elif command.command == 'shutdown':
print("🎙️ 输入进程:收到关闭命令")
self.running = False
return
except queue.Empty:
pass
def _process_audio(self):
"""处理音频数据"""
try:
data = self.input_stream.read(self.CHUNK_SIZE, exception_on_overflow=False)
if len(data) == 0:
return
# 更新预录音缓冲区
self._update_pre_record_buffer(data)
# ZCR语音检测
zcr = self._calculate_zcr(data)
# 语音检测
is_voice = self._is_voice_active(zcr)
if self.is_recording:
# 录音模式
self.recording_buffer.append(data)
# 静音检测
if is_voice:
self.silence_start_time = None
self.consecutive_silence_count = 0
self.consecutive_low_zcr_count = 0 # 重置低ZCR计数
else:
self.consecutive_silence_count += 1
self.consecutive_low_zcr_count += 1
if self.silence_start_time is None:
self.silence_start_time = time.time()
# 检查是否应该停止录音
recording_duration = time.time() - self.recording_start_time
should_stop = False
# ZCR静音检测
if (self.consecutive_low_zcr_count >= self.low_zcr_threshold_count and
recording_duration >= self.config['min_recording_time']):
should_stop = True
print(f"🎙️ 输入进程ZCR静音检测触发停止录音")
# 最大时间检测
if recording_duration >= self.config['max_recording_time']:
should_stop = True
print(f"🎙️ 输入进程:达到最大录音时间")
if should_stop:
self._stop_recording()
else:
# 监听模式
if is_voice:
# 检测到语音,开始录音
self._start_recording()
else:
# 显示监听状态
buffer_usage = len(self.pre_record_buffer) / self.pre_record_max_frames * 100
print(f"\r🎙️ 监听中... ZCR: {zcr:.0f} | 语音: {is_voice} | 缓冲: {buffer_usage:.0f}%", end='', flush=True)
except Exception as e:
print(f"🎙️ 输入进程音频处理错误: {e}")
def _update_pre_record_buffer(self, audio_data: bytes):
"""更新预录音缓冲区"""
self.pre_record_buffer.append(audio_data)
# 保持缓冲区大小
if len(self.pre_record_buffer) > self.pre_record_max_frames:
self.pre_record_buffer.pop(0)
def _start_recording(self):
"""开始录音"""
if not self.recording_enabled:
return
self.is_recording = True
self.recording_buffer = []
self.recording_start_time = time.time()
self.silence_start_time = None
self.consecutive_silence_count = 0
self.consecutive_low_zcr_count = 0
# 将预录音缓冲区的内容添加到录音中
self.recording_buffer.extend(self.pre_record_buffer)
self.pre_record_buffer.clear()
print(f"🎙️ 输入进程:开始录音(包含预录音 {self.config['pre_record_duration']}秒)")
def _stop_recording(self):
"""停止录音并发送数据"""
if not self.is_recording:
return
self.is_recording = False
# 合并录音数据
if self.recording_buffer:
audio_data = b''.join(self.recording_buffer)
duration = len(audio_data) / (self.RATE * 2)
# 创建音频片段
segment = AudioSegment(
audio_data=audio_data,
start_time=self.recording_start_time,
end_time=time.time(),
duration=duration,
metadata={
'sample_rate': self.RATE,
'channels': self.CHANNELS,
'format': self.FORMAT,
'chunk_size': self.CHUNK_SIZE
}
)
# 保存录音文件(可选)
filename = self._save_recording(audio_data)
# 发送给主进程
self.event_queue.put(ProcessEvent(
event_type='recording_complete',
data=audio_data,
metadata={
'duration': duration,
'start_time': self.recording_start_time,
'filename': filename
}
))
print(f"📝 输入进程:录音完成,时长 {duration:.2f}")
# 清空缓冲区
self.recording_buffer = []
self.pre_record_buffer = []
def _save_recording(self, audio_data: bytes) -> str:
"""保存录音文件"""
try:
timestamp = time.strftime("%Y%m%d_%H%M%S")
filename = f"recording_{timestamp}.wav"
with wave.open(filename, 'wb') as wf:
wf.setnchannels(self.CHANNELS)
wf.setsampwidth(self.audio.get_sample_size(self.FORMAT))
wf.setframerate(self.RATE)
wf.writeframes(audio_data)
print(f"💾 输入进程:录音已保存到 {filename}")
return filename
except Exception as e:
print(f"❌ 输入进程保存录音失败: {e}")
return None
def _calculate_zcr(self, audio_data: bytes) -> float:
"""计算零交叉率"""
if len(audio_data) == 0:
return 0
audio_array = np.frombuffer(audio_data, dtype=np.int16)
# 计算零交叉次数
zero_crossings = np.sum(np.diff(np.sign(audio_array)) != 0)
# 归一化到采样率
zcr = zero_crossings / len(audio_array) * self.RATE
# 更新ZCR历史
self.zcr_history.append(zcr)
if len(self.zcr_history) > self.max_zcr_history:
self.zcr_history.pop(0)
return zcr
def _is_voice_active(self, zcr: float) -> bool:
"""基于ZCR判断是否为语音活动"""
# 简单的ZCR范围检测匹配recorder.py的实现
return 2400 < zcr < 12000
def _cleanup(self):
"""清理资源"""
if self.input_stream:
try:
self.input_stream.stop_stream()
self.input_stream.close()
except:
pass
if self.audio:
try:
self.audio.terminate()
except:
pass
class OutputProcess:
"""输出进程 - 专门负责音频播放"""
def __init__(self, audio_queue: mp.Queue, config: Dict[str, Any] = None):
self.audio_queue = audio_queue # 主进程 → 输出进程
self.config = config or self._get_default_config()
# 音频播放参数
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
self.RATE = 16000
self.CHUNK_SIZE = 512
# 播放状态
self.is_playing = False
self.playback_buffer = []
self.total_chunks_played = 0
self.total_audio_size = 0
# PyAudio实例
self.audio = None
self.output_stream = None
# 运行状态
self.running = True
def _get_default_config(self) -> Dict[str, Any]:
"""获取默认配置"""
return {
'buffer_size': 1000,
'show_progress': True,
'progress_interval': 100
}
def run(self):
"""输出进程主循环"""
print("🔊 输出进程启动")
self._setup_audio()
try:
while self.running:
# 处理音频队列
self._process_audio_queue()
# 播放缓冲的音频
self._play_audio()
# 显示播放进度
self._show_progress()
time.sleep(0.001) # 极短休眠,确保流畅播放
except KeyboardInterrupt:
print("🔊 输出进程收到中断信号")
except Exception as e:
print(f"❌ 输出进程错误: {e}")
finally:
self._cleanup()
print("🔊 输出进程退出")
def _setup_audio(self):
"""设置音频输出设备"""
try:
self.audio = pyaudio.PyAudio()
self.output_stream = self.audio.open(
format=self.FORMAT,
channels=self.CHANNELS,
rate=self.RATE,
output=True,
frames_per_buffer=self.CHUNK_SIZE
)
print("🔊 输出进程:音频设备初始化成功")
except Exception as e:
print(f"❌ 输出进程音频设备初始化失败: {e}")
raise
def _process_audio_queue(self):
"""处理来自主进程的音频数据"""
try:
while True:
audio_data = self.audio_queue.get_nowait()
if audio_data is None:
# 结束信号
self._finish_playback()
break
if isinstance(audio_data, str) and audio_data.startswith("METADATA:"):
# 处理元数据
metadata = audio_data[9:] # 移除 "METADATA:" 前缀
print(f"📝 输出进程:播放元数据 {metadata}")
continue
# 音频数据放入播放缓冲区
self.playback_buffer.append(audio_data)
if not self.is_playing:
self.is_playing = True
print("🔊 输出进程:开始播放音频")
except queue.Empty:
pass
def _play_audio(self):
"""播放音频数据"""
if self.playback_buffer and self.output_stream:
try:
# 取出一块音频数据播放
audio_chunk = self.playback_buffer.pop(0)
if audio_chunk and len(audio_chunk) > 0:
self.output_stream.write(audio_chunk)
self.total_chunks_played += 1
self.total_audio_size += len(audio_chunk)
except Exception as e:
print(f"❌ 输出进程播放错误: {e}")
self.playback_buffer.clear()
def _show_progress(self):
"""显示播放进度"""
if (self.config['show_progress'] and
self.total_chunks_played > 0 and
self.total_chunks_played % self.config['progress_interval'] == 0):
progress = f"🔊 播放进度: {self.total_chunks_played} 块 | {self.total_audio_size / 1024:.1f} KB"
print(f"\r{progress}", end='', flush=True)
def _finish_playback(self):
"""完成播放"""
self.is_playing = False
self.playback_buffer.clear()
if self.total_chunks_played > 0:
print(f"\n✅ 输出进程:播放完成,总计 {self.total_chunks_played} 块, {self.total_audio_size / 1024:.1f} KB")
# 重置统计
self.total_chunks_played = 0
self.total_audio_size = 0
# 通知主进程播放完成
# 这里可以通过共享内存或另一个队列来实现
# 暂时简化处理,由主进程通过队列大小判断
def _cleanup(self):
"""清理资源"""
if self.output_stream:
try:
self.output_stream.stop_stream()
self.output_stream.close()
except:
pass
if self.audio:
try:
self.audio.terminate()
except:
pass
if __name__ == "__main__":
# 测试代码
print("音频进程模块测试")
print("这个模块应该在多进程环境中运行")

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{
"system": {
"max_queue_size": 1000,
"process_timeout": 30,
"heartbeat_interval": 1.0,
"log_level": "INFO"
},
"audio": {
"sample_rate": 16000,
"channels": 1,
"chunk_size": 1024,
"format": "paInt16"
},
"recording": {
"min_duration": 3.0,
"max_duration": 30.0,
"silence_threshold": 3.0,
"pre_record_duration": 2.0
},
"processing": {
"enable_asr": true,
"enable_llm": true,
"enable_tts": true,
"character": "libai",
"max_tokens": 50
},
"detection": {
"zcr_min": 2400,
"zcr_max": 12000,
"consecutive_silence_count": 30,
"max_zcr_history": 50
},
"playback": {
"buffer_size": 1000,
"show_progress": true,
"progress_interval": 100,
"chunk_size": 512
}
}

774
control_system.py Normal file
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@ -0,0 +1,774 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
多进程音频控制系统
实现主控制进程和状态管理
"""
import multiprocessing as mp
import queue
import time
import threading
import requests
import json
import base64
import gzip
import uuid
import asyncio
import websockets
from typing import Optional, Dict, Any, List
from dataclasses import dataclass, asdict
from enum import Enum
import os
import sys
from audio_processes import (
InputProcess, OutputProcess,
RecordingState, ControlCommand, ProcessEvent
)
class ControlSystem:
"""主控制系统"""
def __init__(self, config: Dict[str, Any] = None):
self.config = config or self._get_default_config()
# 进程间通信
self.input_command_queue = mp.Queue(maxsize=100) # 主进程 → 输入进程
self.input_event_queue = mp.Queue(maxsize=100) # 输入进程 → 主进程
self.output_audio_queue = mp.Queue(maxsize=1000) # 主进程 → 输出进程
# 进程
self.input_process = None
self.output_process = None
# 状态管理
self.state = RecordingState.IDLE
self.processing_complete = False
self.playback_complete = False
# 当前处理的数据
self.current_audio_data = None
self.current_audio_metadata = None
# API配置
self.api_config = self._setup_api_config()
# 统计信息
self.stats = {
'total_conversations': 0,
'total_recording_time': 0,
'successful_processing': 0,
'failed_processing': 0
}
# 运行状态
self.running = True
# 检查依赖
self._check_dependencies()
def _get_default_config(self) -> Dict[str, Any]:
"""获取默认配置"""
return {
'system': {
'max_queue_size': 1000,
'process_timeout': 30,
'heartbeat_interval': 1.0
},
'audio': {
'sample_rate': 16000,
'channels': 1,
'chunk_size': 1024
},
'recording': {
'min_duration': 2.0,
'max_duration': 30.0,
'silence_threshold': 3.0
},
'processing': {
'enable_asr': True,
'enable_llm': True,
'enable_tts': True,
'character': 'libai'
}
}
def _setup_api_config(self) -> Dict[str, Any]:
"""设置API配置"""
config = {
'asr': {
'appid': "8718217928",
'token': "ynJMX-5ix1FsJvswC9KTNlGUdubcchqc",
'cluster': "volcengine_input_common",
'ws_url': "wss://openspeech.bytedance.com/api/v2/asr"
},
'llm': {
'api_url': "https://ark.cn-beijing.volces.com/api/v3/chat/completions",
'model': "doubao-seed-1-6-flash-250828",
'api_key': os.environ.get("ARK_API_KEY", ""),
'max_tokens': 50
},
'tts': {
'url': "https://openspeech.bytedance.com/api/v3/tts/unidirectional",
'app_id': "8718217928",
'access_key': "ynJMX-5ix1FsJvswC9KTNlGUdubcchqc",
'resource_id': "volc.service_type.10029",
'app_key': "aGjiRDfUWi",
'speaker': "zh_female_wanqudashu_moon_bigtts"
}
}
# 加载角色配置
character_config = self._load_character_config(self.config['processing']['character'])
if character_config and "voice" in character_config:
config['tts']['speaker'] = character_config["voice"]
return config
def _load_character_config(self, character_name: str) -> Optional[Dict[str, Any]]:
"""加载角色配置"""
characters_dir = os.path.join(os.path.dirname(__file__), "characters")
config_file = os.path.join(characters_dir, f"{character_name}.json")
if not os.path.exists(config_file):
print(f"⚠️ 角色配置文件不存在: {config_file}")
return None
try:
with open(config_file, 'r', encoding='utf-8') as f:
config = json.load(f)
print(f"✅ 加载角色: {config.get('name', character_name)}")
return config
except Exception as e:
print(f"❌ 加载角色配置失败: {e}")
return None
def _check_dependencies(self):
"""检查依赖库"""
missing_deps = []
try:
import pyaudio
except ImportError:
missing_deps.append("pyaudio")
try:
import numpy
except ImportError:
missing_deps.append("numpy")
try:
import requests
except ImportError:
missing_deps.append("requests")
try:
import websockets
except ImportError:
missing_deps.append("websockets")
if missing_deps:
print(f"❌ 缺少依赖库: {', '.join(missing_deps)}")
print("请安装: pip install " + " ".join(missing_deps))
sys.exit(1)
# 检查API密钥
if not self.api_config['llm']['api_key']:
print("⚠️ 未设置 ARK_API_KEY 环境变量,大语言模型功能将被禁用")
self.config['processing']['enable_llm'] = False
def start(self):
"""启动系统"""
print("🚀 启动多进程音频控制系统")
print("=" * 60)
# 创建并启动输入进程
input_config = {
'zcr_min': 2400,
'zcr_max': 12000,
'min_recording_time': self.config['recording']['min_duration'],
'max_recording_time': self.config['recording']['max_duration'],
'silence_threshold': self.config['recording']['silence_threshold'],
'pre_record_duration': 2.0
}
self.input_process = mp.Process(
target=InputProcess(
self.input_command_queue,
self.input_event_queue,
input_config
).run
)
# 创建并启动输出进程
output_config = {
'buffer_size': 1000,
'show_progress': True,
'progress_interval': 100
}
self.output_process = mp.Process(
target=OutputProcess(
self.output_audio_queue,
output_config
).run
)
# 启动进程
self.input_process.start()
self.output_process.start()
print("✅ 所有进程已启动")
print("🎙️ 输入进程:负责录音和语音检测")
print("🔊 输出进程:负责音频播放")
print("🎯 主控制负责协调和AI处理")
print("=" * 60)
# 启动主控制循环
self._control_loop()
def _control_loop(self):
"""主控制循环"""
print("🎯 主控制循环启动")
try:
while self.running:
# 根据状态处理不同逻辑
if self.state == RecordingState.IDLE:
self._handle_idle_state()
elif self.state == RecordingState.RECORDING:
self._handle_recording_state()
elif self.state == RecordingState.PROCESSING:
self._handle_processing_state()
elif self.state == RecordingState.PLAYING:
self._handle_playing_state()
# 检查进程事件
self._check_events()
# 显示状态
self._display_status()
# 控制循环频率
time.sleep(0.1)
except KeyboardInterrupt:
print("\n👋 收到退出信号...")
self.shutdown()
except Exception as e:
print(f"❌ 主控制循环错误: {e}")
self.shutdown()
def _handle_idle_state(self):
"""处理空闲状态"""
if self.state == RecordingState.IDLE:
# 启用输入进程录音功能
self.input_command_queue.put(ControlCommand('enable_recording'))
self.state = RecordingState.RECORDING
print("🎯 状态IDLE → RECORDING")
def _handle_recording_state(self):
"""处理录音状态"""
# 等待输入进程发送录音完成事件
pass
def _handle_processing_state(self):
"""处理状态"""
if not self.processing_complete:
self._process_audio_pipeline()
def _handle_playing_state(self):
"""处理播放状态"""
# 检查播放是否完成
if self.output_audio_queue.qsize() == 0 and not self.playback_complete:
# 等待一小段时间确保播放完成
time.sleep(0.5)
if self.output_audio_queue.qsize() == 0:
self.playback_complete = True
self.stats['total_conversations'] += 1
def _check_events(self):
"""检查进程事件"""
# 检查输入进程事件
try:
while True:
event = self.input_event_queue.get_nowait()
if event.event_type == 'recording_complete':
print("📡 主控制:收到录音完成事件")
self._handle_recording_complete(event)
except queue.Empty:
pass
def _handle_recording_complete(self, event: ProcessEvent):
"""处理录音完成事件"""
# 禁用输入进程录音功能
self.input_command_queue.put(ControlCommand('disable_recording'))
# 保存录音数据
self.current_audio_data = event.data
self.current_audio_metadata = event.metadata
# 更新统计
self.stats['total_recording_time'] += event.metadata['duration']
# 切换到处理状态
self.state = RecordingState.PROCESSING
self.processing_complete = False
self.playback_complete = False
print(f"🎯 状态RECORDING → PROCESSING (时长: {event.metadata['duration']:.2f}s)")
def _process_audio_pipeline(self):
"""处理音频流水线STT + LLM + TTS"""
try:
print("🤖 开始处理音频流水线")
# 1. 语音识别 (STT)
if self.config['processing']['enable_asr']:
text = self._speech_to_text(self.current_audio_data)
if not text:
print("❌ 语音识别失败")
self._handle_processing_failure()
return
print(f"📝 识别结果: {text}")
else:
text = "语音识别功能已禁用"
# 2. 大语言模型 (LLM)
if self.config['processing']['enable_llm']:
response = self._call_llm(text)
if not response:
print("❌ 大语言模型调用失败")
self._handle_processing_failure()
return
print(f"💬 AI回复: {response}")
else:
response = "大语言模型功能已禁用"
# 3. 文本转语音 (TTS)
if self.config['processing']['enable_tts']:
success = self._text_to_speech_streaming(response)
if not success:
print("❌ 文本转语音失败")
self._handle_processing_failure()
return
else:
print(" 文本转语音功能已禁用")
# 直接发送结束信号
self.output_audio_queue.put(None)
# 标记处理完成
self.processing_complete = True
self.state = RecordingState.PLAYING
self.stats['successful_processing'] += 1
print("🎯 状态PROCESSING → PLAYING")
except Exception as e:
print(f"❌ 处理流水线错误: {e}")
self._handle_processing_failure()
def _handle_processing_failure(self):
"""处理失败情况"""
self.stats['failed_processing'] += 1
self.state = RecordingState.IDLE
self.processing_complete = True
self.playback_complete = True
print("🎯 状态PROCESSING → IDLE (失败)")
def _speech_to_text(self, audio_data: bytes) -> Optional[str]:
"""语音转文字"""
try:
return asyncio.run(self._recognize_audio_async(audio_data))
except Exception as e:
print(f"❌ 语音识别异常: {e}")
return None
async def _recognize_audio_async(self, audio_data: bytes) -> Optional[str]:
"""异步语音识别"""
if not self.config['processing']['enable_asr']:
return "语音识别功能已禁用"
try:
import websockets
# 生成ASR头部
def generate_asr_header(message_type=1, message_type_specific_flags=0):
PROTOCOL_VERSION = 0b0001
DEFAULT_HEADER_SIZE = 0b0001
JSON = 0b0001
GZIP = 0b0001
header = bytearray()
header.append((PROTOCOL_VERSION << 4) | DEFAULT_HEADER_SIZE)
header.append((message_type << 4) | message_type_specific_flags)
header.append((JSON << 4) | GZIP)
header.append(0x00) # reserved
return header
# 解析ASR响应
def parse_asr_response(res):
# 简化的响应解析
if len(res) < 8:
return {}
message_type = res[1] >> 4
payload_size = int.from_bytes(res[4:8], "big", signed=False)
payload_msg = res[8:8+payload_size]
if message_type == 0b1001: # SERVER_FULL_RESPONSE
try:
if payload_msg.startswith(b'{'):
result = json.loads(payload_msg.decode('utf-8'))
return result
except:
pass
return {}
# 构建请求参数
reqid = str(uuid.uuid4())
request_params = {
'app': {
'appid': self.api_config['asr']['appid'],
'cluster': self.api_config['asr']['cluster'],
'token': self.api_config['asr']['token'],
},
'user': {
'uid': 'multiprocess_asr'
},
'request': {
'reqid': reqid,
'nbest': 1,
'workflow': 'audio_in,resample,partition,vad,fe,decode,itn,nlu_punctuate',
'show_language': False,
'show_utterances': False,
'result_type': 'full',
"sequence": 1
},
'audio': {
'format': 'wav',
'rate': self.config['audio']['sample_rate'],
'language': 'zh-CN',
'bits': 16,
'channel': self.config['audio']['channels'],
'codec': 'raw'
}
}
# 构建请求
payload_bytes = str.encode(json.dumps(request_params))
payload_bytes = gzip.compress(payload_bytes)
full_client_request = bytearray(generate_asr_header())
full_client_request.extend((len(payload_bytes)).to_bytes(4, 'big'))
full_client_request.extend(payload_bytes)
# 设置认证头
additional_headers = {'Authorization': 'Bearer; {}'.format(self.api_config['asr']['token'])}
# 连接WebSocket
async with websockets.connect(
self.api_config['asr']['ws_url'],
additional_headers=additional_headers,
max_size=1000000000
) as ws:
# 发送请求
await ws.send(full_client_request)
res = await ws.recv()
result = parse_asr_response(res)
# 发送音频数据
chunk_size = int(self.config['audio']['channels'] * 2 *
self.config['audio']['sample_rate'] * 15000 / 1000)
for offset in range(0, len(audio_data), chunk_size):
chunk = audio_data[offset:offset + chunk_size]
last = (offset + chunk_size) >= len(audio_data)
payload_bytes = gzip.compress(chunk)
audio_only_request = bytearray(
generate_asr_header(
message_type=0b0010,
message_type_specific_flags=0b0010 if last else 0
)
)
audio_only_request.extend((len(payload_bytes)).to_bytes(4, 'big'))
audio_only_request.extend(payload_bytes)
await ws.send(audio_only_request)
res = await ws.recv()
result = parse_asr_response(res)
# 获取最终结果
if 'payload_msg' in result and 'result' in result['payload_msg']:
results = result['payload_msg']['result']
if results:
return results[0].get('text', '识别失败')
return None
except Exception as e:
print(f"❌ 语音识别失败: {e}")
return None
def _call_llm(self, text: str) -> Optional[str]:
"""调用大语言模型"""
if not self.config['processing']['enable_llm']:
return "大语言模型功能已禁用"
try:
# 获取角色配置
character_config = self._load_character_config(self.config['processing']['character'])
if character_config and "system_prompt" in character_config:
system_prompt = character_config["system_prompt"]
else:
system_prompt = "你是一个智能助手,请根据用户的语音输入提供有帮助的回答。保持回答简洁明了。"
# 构建请求
headers = {
"Content-Type": "application/json",
"Authorization": f"Bearer {self.api_config['llm']['api_key']}"
}
messages = [
{"role": "system", "content": system_prompt},
{"role": "user", "content": text}
]
data = {
"model": self.api_config['llm']['model'],
"messages": messages,
"max_tokens": self.api_config['llm']['max_tokens'],
"stream": False # 非流式,简化实现
}
response = requests.post(
self.api_config['llm']['api_url'],
headers=headers,
json=data,
timeout=30
)
if response.status_code == 200:
result = response.json()
if 'choices' in result and len(result['choices']) > 0:
content = result['choices'][0]['message']['content']
return content.strip()
print(f"❌ LLM API调用失败: {response.status_code}")
return None
except Exception as e:
print(f"❌ 大语言模型调用失败: {e}")
return None
def _text_to_speech_streaming(self, text: str) -> bool:
"""文本转语音(流式)"""
if not self.config['processing']['enable_tts']:
return False
try:
print("🎵 开始文本转语音")
# 发送元数据
self.output_audio_queue.put(f"METADATA:{text[:30]}...")
# 构建请求头
headers = {
"X-Api-App-Id": self.api_config['tts']['app_id'],
"X-Api-Access-Key": self.api_config['tts']['access_key'],
"X-Api-Resource-Id": self.api_config['tts']['resource_id'],
"X-Api-App-Key": self.api_config['tts']['app_key'],
"Content-Type": "application/json",
"Connection": "keep-alive"
}
# 构建请求参数
payload = {
"user": {
"uid": "multiprocess_tts"
},
"req_params": {
"text": text,
"speaker": self.api_config['tts']['speaker'],
"audio_params": {
"format": "pcm",
"sample_rate": self.config['audio']['sample_rate'],
"enable_timestamp": True
},
"additions": "{\"explicit_language\":\"zh\",\"disable_markdown_filter\":true, \"enable_timestamp\":true}\"}"
}
}
# 发送请求
session = requests.Session()
try:
response = session.post(
self.api_config['tts']['url'],
headers=headers,
json=payload,
stream=True
)
if response.status_code != 200:
print(f"❌ TTS请求失败: {response.status_code}")
return False
# 处理流式响应
total_audio_size = 0
chunk_count = 0
for chunk in response.iter_lines(decode_unicode=True):
if not chunk:
continue
try:
data = json.loads(chunk)
if data.get("code", 0) == 0 and "data" in data and data["data"]:
chunk_audio = base64.b64decode(data["data"])
audio_size = len(chunk_audio)
total_audio_size += audio_size
chunk_count += 1
# 发送到输出进程
self.output_audio_queue.put(chunk_audio)
# 显示进度
if chunk_count % 10 == 0:
progress = f"📥 TTS生成: {chunk_count} 块 | {total_audio_size / 1024:.1f} KB"
print(f"\r{progress}", end='', flush=True)
if data.get("code", 0) == 20000000:
break
except json.JSONDecodeError:
continue
print(f"\n✅ TTS音频生成完成: {chunk_count} 块, {total_audio_size / 1024:.1f} KB")
# 发送结束信号
self.output_audio_queue.put(None)
return chunk_count > 0
finally:
response.close()
session.close()
except Exception as e:
print(f"❌ 文本转语音失败: {e}")
return False
def _display_status(self):
"""显示系统状态"""
# 每秒显示一次状态
if hasattr(self, '_last_status_time'):
if time.time() - self._last_status_time < 1.0:
return
self._last_status_time = time.time()
# 状态显示
status_lines = [
f"🎯 状态: {self.state.value}",
f"📊 统计: 对话{self.stats['total_conversations']} | "
f"录音{self.stats['total_recording_time']:.1f}s | "
f"成功{self.stats['successful_processing']} | "
f"失败{self.stats['failed_processing']}"
]
# 队列状态
input_queue_size = self.input_command_queue.qsize()
output_queue_size = self.output_audio_queue.qsize()
if input_queue_size > 0 or output_queue_size > 0:
status_lines.append(f"📦 队列: 输入{input_queue_size} | 输出{output_queue_size}")
# 显示状态
status_str = " | ".join(status_lines)
print(f"\r{status_str}", end='', flush=True)
def shutdown(self):
"""关闭系统"""
print("\n🛑 正在关闭系统...")
self.running = False
# 发送关闭命令
try:
self.input_command_queue.put(ControlCommand('shutdown'))
self.output_audio_queue.put(None)
except:
pass
# 等待进程结束
if self.input_process:
try:
self.input_process.join(timeout=5)
except:
pass
if self.output_process:
try:
self.output_process.join(timeout=5)
except:
pass
# 显示最终统计
print("\n📊 最终统计:")
print(f" 总对话次数: {self.stats['total_conversations']}")
print(f" 总录音时长: {self.stats['total_recording_time']:.1f}")
print(f" 成功处理: {self.stats['successful_processing']}")
print(f" 失败处理: {self.stats['failed_processing']}")
success_rate = (self.stats['successful_processing'] /
max(1, self.stats['successful_processing'] + self.stats['failed_processing']) * 100)
print(f" 成功率: {success_rate:.1f}%")
print("👋 系统已关闭")
def main():
"""主函数"""
import argparse
parser = argparse.ArgumentParser(description='多进程音频控制系统')
parser.add_argument('--character', '-c', type=str, default='libai',
help='选择角色 (默认: libai)')
parser.add_argument('--config', type=str,
help='配置文件路径')
args = parser.parse_args()
# 加载配置
config = None
if args.config:
try:
with open(args.config, 'r', encoding='utf-8') as f:
config = json.load(f)
except Exception as e:
print(f"⚠️ 配置文件加载失败: {e}")
# 创建控制系统
control_system = ControlSystem(config)
# 设置角色
if args.character:
control_system.config['processing']['character'] = args.character
# 启动系统
control_system.start()
if __name__ == "__main__":
main()

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@ -1,74 +0,0 @@
#!/bin/bash
# 智能语音助手系统安装脚本
# 适用于树莓派和Linux系统
echo "🚀 智能语音助手系统 - 安装脚本"
echo "================================"
# 检查是否为root用户
if [ "$EUID" -eq 0 ]; then
echo "⚠️ 请不要以root身份运行此脚本"
echo " 建议使用普通用户: sudo ./install.sh"
exit 1
fi
# 更新包管理器
echo "📦 更新包管理器..."
sudo apt-get update
# 安装系统依赖
echo "🔧 安装系统依赖..."
sudo apt-get install -y \
python3 \
python3-pip \
portaudio19-dev \
python3-dev \
alsa-utils
# 安装Python依赖
echo "🐍 安装Python依赖..."
pip3 install --user \
websockets \
requests \
pyaudio \
numpy
# 检查音频播放器
echo "🔊 检查音频播放器..."
if command -v aplay >/dev/null 2>&1; then
echo "✅ aplay 已安装支持PCM/WAV播放"
else
echo "❌ aplay 安装失败"
fi
# 检查Python模块
echo "🧪 检查Python模块..."
python3 -c "import websockets, requests, pyaudio, numpy" 2>/dev/null
if [ $? -eq 0 ]; then
echo "✅ 所有Python依赖已安装"
else
echo "❌ 部分Python依赖安装失败"
fi
echo ""
echo "✅ 安装完成!"
echo ""
echo "📋 使用说明:"
echo "1. 设置API密钥如需使用大语言模型:"
echo " export ARK_API_KEY='your_api_key_here'"
echo ""
echo "2. 运行程序:"
echo " python3 recorder.py"
echo ""
echo "3. 故障排除:"
echo " - 如果遇到权限问题请确保用户在audio组中:"
echo " sudo usermod -a -G audio \$USER"
echo " - 然后重新登录或重启系统"
echo ""
echo "🎯 系统功能:"
echo "- 🎙️ 智能语音录制"
echo "- 🤖 在线语音识别"
echo "- 💬 AI智能对话"
echo "- 🔊 语音回复合成"
echo "- 📁 自动文件管理"

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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
多进程音频录音系统
基于进程隔离的音频处理架构
"""
import os
import sys
import argparse
import json
import time
from typing import Dict, Any
def check_dependencies():
"""检查系统依赖"""
missing_deps = []
try:
import pyaudio
except ImportError:
missing_deps.append("pyaudio")
try:
import numpy
except ImportError:
missing_deps.append("numpy")
try:
import requests
except ImportError:
missing_deps.append("requests")
try:
import websockets
except ImportError:
missing_deps.append("websockets")
if missing_deps:
print("❌ 缺少以下依赖库:")
for dep in missing_deps:
print(f" - {dep}")
print("\n请运行以下命令安装:")
print(f"pip install {' '.join(missing_deps)}")
return False
return True
def check_environment():
"""检查运行环境"""
print("🔍 检查运行环境...")
# 检查Python版本
python_version = sys.version_info
if python_version.major < 3 or (python_version.major == 3 and python_version.minor < 7):
print(f"❌ Python版本过低: {python_version.major}.{python_version.minor}")
print("需要Python 3.7或更高版本")
return False
print(f"✅ Python版本: {python_version.major}.{python_version.minor}.{python_version.micro}")
# 检查操作系统
import platform
system = platform.system().lower()
print(f"✅ 操作系统: {system}")
# 检查音频设备
try:
import pyaudio
audio = pyaudio.PyAudio()
device_count = audio.get_device_count()
print(f"✅ 音频设备数量: {device_count}")
if device_count == 0:
print("❌ 未检测到音频设备")
return False
audio.terminate()
except Exception as e:
print(f"❌ 音频设备检查失败: {e}")
return False
# 检查网络连接
try:
import requests
response = requests.get("https://www.baidu.com", timeout=5)
print("✅ 网络连接正常")
except:
print("⚠️ 网络连接可能有问题会影响在线AI功能")
# 检查API密钥
api_key = os.environ.get("ARK_API_KEY")
if api_key:
print("✅ ARK_API_KEY 已设置")
else:
print("⚠️ ARK_API_KEY 未设置,大语言模型功能将被禁用")
print(" 请运行: export ARK_API_KEY='your_api_key_here'")
return True
def list_characters():
"""列出可用角色"""
characters_dir = os.path.join(os.path.dirname(__file__), "characters")
if not os.path.exists(characters_dir):
print("❌ 角色目录不存在")
return
characters = []
for file in os.listdir(characters_dir):
if file.endswith('.json'):
character_name = file[:-5]
config_file = os.path.join(characters_dir, file)
try:
with open(config_file, 'r', encoding='utf-8') as f:
config = json.load(f)
name = config.get('name', character_name)
desc = config.get('description', '无描述')
characters.append(f"{character_name}: {name} - {desc}")
except:
characters.append(f"{character_name}: 配置文件读取失败")
if characters:
print("🎭 可用角色列表:")
for char in characters:
print(f" - {char}")
else:
print("❌ 未找到任何角色配置文件")
def create_sample_config():
"""创建示例配置文件"""
config = {
"system": {
"max_queue_size": 1000,
"process_timeout": 30,
"heartbeat_interval": 1.0,
"log_level": "INFO"
},
"audio": {
"sample_rate": 16000,
"channels": 1,
"chunk_size": 1024,
"format": "paInt16"
},
"recording": {
"min_duration": 2.0,
"max_duration": 30.0,
"silence_threshold": 3.0,
"pre_record_duration": 2.0
},
"processing": {
"enable_asr": True,
"enable_llm": True,
"enable_tts": True,
"character": "libai",
"max_tokens": 50
},
"detection": {
"zcr_min": 2400,
"zcr_max": 12000,
"consecutive_silence_count": 30,
"max_zcr_history": 30
},
"playback": {
"buffer_size": 1000,
"show_progress": True,
"progress_interval": 100,
"chunk_size": 512
}
}
config_file = "config.json"
try:
with open(config_file, 'w', encoding='utf-8') as f:
json.dump(config, f, indent=2, ensure_ascii=False)
print(f"✅ 示例配置文件已创建: {config_file}")
except Exception as e:
print(f"❌ 创建配置文件失败: {e}")
def main():
"""主函数"""
parser = argparse.ArgumentParser(
description='多进程音频录音系统',
formatter_class=argparse.RawDescriptionHelpFormatter,
epilog="""
使用示例:
python multiprocess_recorder.py # 使用默认角色
python multiprocess_recorder.py -c zhubajie # 指定角色
python multiprocess_recorder.py -l # 列出角色
python multiprocess_recorder.py --create-config # 创建配置文件
"""
)
parser.add_argument('--character', '-c', type=str, default='libai',
help='选择角色 (默认: libai)')
parser.add_argument('--list-characters', '-l', action='store_true',
help='列出所有可用角色')
parser.add_argument('--config', type=str,
help='配置文件路径')
parser.add_argument('--create-config', action='store_true',
help='创建示例配置文件')
parser.add_argument('--check-env', action='store_true',
help='检查运行环境')
parser.add_argument('--verbose', '-v', action='store_true',
help='详细输出')
args = parser.parse_args()
# 显示欢迎信息
print("🚀 多进程音频录音系统")
print("=" * 60)
# 检查依赖
if not check_dependencies():
sys.exit(1)
# 创建配置文件
if args.create_config:
create_sample_config()
return
# 检查环境
if args.check_env:
check_environment()
return
# 列出角色
if args.list_characters:
list_characters()
return
# 检查characters目录
characters_dir = os.path.join(os.path.dirname(__file__), "characters")
if not os.path.exists(characters_dir):
print(f"⚠️ 角色目录不存在: {characters_dir}")
print("请确保characters目录存在并包含角色配置文件")
# 检查指定角色
character_file = os.path.join(characters_dir, f"{args.character}.json")
if not os.path.exists(character_file):
print(f"⚠️ 角色文件不存在: {character_file}")
print(f"可用角色:")
list_characters()
return
print(f"🎭 当前角色: {args.character}")
print("🎯 系统特点:")
print(" - 多进程架构:输入输出完全隔离")
print(" - 零切换延迟:无需音频设备重置")
print(" - 实时响应:并行处理录音和播放")
print(" - 智能检测基于ZCR的语音识别")
print(" - 流式TTS实时音频生成和播放")
print(" - 角色扮演支持多种AI角色")
print("=" * 60)
# 显示使用说明
print("📖 使用说明:")
print(" - 检测到语音自动开始录音")
print(" - 持续静音3秒自动结束录音")
print(" - 录音完成后自动处理和播放")
print(" - 按 Ctrl+C 退出")
print("=" * 60)
# 加载配置
config = None
if args.config:
try:
with open(args.config, 'r', encoding='utf-8') as f:
config = json.load(f)
print(f"📋 加载配置文件: {args.config}")
except Exception as e:
print(f"⚠️ 配置文件加载失败: {e}")
print("使用默认配置")
try:
# 导入控制系统
from control_system import ControlSystem
# 创建控制系统
control_system = ControlSystem(config)
# 设置角色
control_system.config['processing']['character'] = args.character
# 设置日志级别
if args.verbose:
control_system.config['system']['log_level'] = "DEBUG"
# 启动系统
control_system.start()
except KeyboardInterrupt:
print("\n👋 用户中断")
except Exception as e:
print(f"❌ 系统启动失败: {e}")
if args.verbose:
import traceback
traceback.print_exc()
finally:
print("👋 系统退出")
if __name__ == "__main__":
main()

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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
快速测试脚本
用于验证多进程录音系统的基础功能
"""
import time
import multiprocessing as mp
from audio_processes import InputProcess, OutputProcess
def test_audio_processes():
"""测试音频进程类"""
print("🧪 测试音频进程类...")
# 创建测试队列
command_queue = mp.Queue()
event_queue = mp.Queue()
audio_queue = mp.Queue()
# 创建进程配置
config = {
'zcr_min': 3000,
'zcr_max': 10000,
'min_recording_time': 3.0,
'max_recording_time': 10.0, # 缩短测试时间
'silence_threshold': 3.0,
'pre_record_duration': 2.0,
'voice_activation_threshold': 5, # 降低阈值便于测试
'calibration_samples': 50, # 减少校准时间
'adaptive_threshold': True
}
# 创建输入进程
input_process = InputProcess(command_queue, event_queue, config)
# 创建输出进程
output_process = OutputProcess(audio_queue)
print("✅ 音频进程类创建成功")
# 测试配置加载
print("📋 测试配置:")
print(f" ZCR范围: {config['zcr_min']} - {config['zcr_max']}")
print(f" 校准样本数: {config['calibration_samples']}")
print(f" 语音激活阈值: {config['voice_activation_threshold']}")
return True
def test_dependencies():
"""测试依赖库"""
print("🔍 检查依赖库...")
dependencies = {
'numpy': False,
'pyaudio': False,
'requests': False,
'websockets': False
}
try:
import numpy
dependencies['numpy'] = True
print("✅ numpy")
except ImportError:
print("❌ numpy")
try:
import pyaudio
dependencies['pyaudio'] = True
print("✅ pyaudio")
except ImportError:
print("❌ pyaudio")
try:
import requests
dependencies['requests'] = True
print("✅ requests")
except ImportError:
print("❌ requests")
try:
import websockets
dependencies['websockets'] = True
print("✅ websockets")
except ImportError:
print("❌ websockets")
missing = [dep for dep, installed in dependencies.items() if not installed]
if missing:
print(f"❌ 缺少依赖: {', '.join(missing)}")
return False
else:
print("✅ 所有依赖都已安装")
return True
def main():
"""主测试函数"""
print("🚀 多进程录音系统快速测试")
print("=" * 50)
# 测试依赖
if not test_dependencies():
print("❌ 依赖检查失败")
return False
print()
# 测试音频进程
if not test_audio_processes():
print("❌ 音频进程测试失败")
return False
print()
print("✅ 所有测试通过!")
print("💡 现在可以运行主程序:")
print(" python multiprocess_recorder.py")
return True
if __name__ == "__main__":
main()

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@ -1,96 +0,0 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
测试大语言模型API功能
"""
import os
import requests
import json
def test_llm_api():
"""测试大语言模型API"""
# 检查API密钥
api_key = os.environ.get("ARK_API_KEY")
if not api_key:
print("❌ 未设置 ARK_API_KEY 环境变量")
return False
print(f"✅ API密钥已设置: {api_key[:20]}...")
# API配置
api_url = "https://ark.cn-beijing.volces.com/api/v3/chat/completions"
model = "doubao-1-5-pro-32k-250115"
# 测试消息
test_message = "你好,请简单介绍一下自己"
try:
print("🤖 测试大语言模型API...")
headers = {
"Content-Type": "application/json",
"Authorization": f"Bearer {api_key}"
}
data = {
"model": model,
"messages": [
{
"role": "system",
"content": "你是一个智能助手,请根据用户的语音输入提供有帮助的回答。保持回答简洁明了。"
},
{
"role": "user",
"content": test_message
}
]
}
response = requests.post(api_url, headers=headers, json=data, timeout=30)
print(f"📡 HTTP状态码: {response.status_code}")
if response.status_code == 200:
result = response.json()
print("✅ API调用成功")
if "choices" in result and len(result["choices"]) > 0:
llm_response = result["choices"][0]["message"]["content"]
print(f"💬 AI回复: {llm_response}")
# 显示完整响应结构
print("\n📋 完整响应结构:")
print(json.dumps(result, indent=2, ensure_ascii=False))
return True
else:
print("❌ 响应格式错误")
print(f"响应内容: {response.text}")
return False
else:
print(f"❌ API调用失败: {response.status_code}")
print(f"响应内容: {response.text}")
return False
except requests.exceptions.RequestException as e:
print(f"❌ 网络请求失败: {e}")
return False
except Exception as e:
print(f"❌ 测试失败: {e}")
return False
if __name__ == "__main__":
print("🧪 测试大语言模型API功能")
print("=" * 50)
success = test_llm_api()
if success:
print("\n✅ 大语言模型功能测试通过!")
print("🚀 现在可以运行完整的语音助手系统了")
else:
print("\n❌ 大语言模型功能测试失败")
print("🔧 请检查API密钥和网络连接")

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@ -1,108 +0,0 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
测试流式响应解析的脚本
"""
import json
import requests
import os
def test_streaming_response():
"""测试流式响应解析"""
# 检查API密钥
api_key = os.environ.get("ARK_API_KEY")
if not api_key:
print("❌ 请设置 ARK_API_KEY 环境变量")
return
headers = {
"Content-Type": "application/json",
"Authorization": f"Bearer {api_key}"
}
data = {
"messages": [
{
"content": "你是一个智能助手,请回答问题。",
"role": "system"
},
{
"content": "你好,请简单介绍一下自己",
"role": "user"
}
],
"model": "doubao-1-5-pro-32k-250115",
"stream": True
}
print("🚀 开始测试流式响应...")
try:
response = requests.post(
"https://ark.cn-beijing.volces.com/api/v3/chat/completions",
headers=headers,
json=data,
stream=True,
timeout=30
)
print(f"📊 响应状态: {response.status_code}")
if response.status_code != 200:
print(f"❌ 请求失败: {response.text}")
return
print("🔍 开始解析流式响应...")
accumulated_text = ""
line_count = 0
for line in response.iter_lines(decode_unicode=True):
line_count += 1
if not line or not line.strip():
continue
# 预处理
line = line.strip()
print(f"\n--- 第{line_count}行 ---")
print(f"原始内容: {repr(line)}")
if line.startswith("data: "):
data_str = line[6:] # 移除 "data: " 前缀
print(f"处理后: {repr(data_str)}")
if data_str == "[DONE]":
print("✅ 流结束")
break
try:
chunk_data = json.loads(data_str)
print(f"✅ JSON解析成功: {chunk_data}")
if "choices" in chunk_data and len(chunk_data["choices"]) > 0:
delta = chunk_data["choices"][0].get("delta", {})
content = delta.get("content", "")
if content:
accumulated_text += content
print(f"💬 累计内容: {accumulated_text}")
except json.JSONDecodeError as e:
print(f"❌ JSON解析失败: {e}")
print(f"🔍 问题数据: {repr(data_str)}")
except Exception as e:
print(f"❌ 其他错误: {e}")
print(f"\n✅ 测试完成,总共处理了 {line_count}")
print(f"📝 最终内容: {accumulated_text}")
except Exception as e:
print(f"❌ 测试失败: {e}")
if __name__ == "__main__":
test_streaming_response()

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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
语音检测测试脚本
用于测试和调试ZCR语音检测功能
"""
import numpy as np
import time
import pyaudio
from audio_processes import InputProcess
import multiprocessing as mp
import queue
class VoiceDetectionTester:
"""语音检测测试器"""
def __init__(self):
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
self.RATE = 16000
self.CHUNK_SIZE = 1024
# 测试参数
self.test_duration = 30 # 测试30秒
self.zcr_history = []
self.voice_count = 0
# 音频设备
self.audio = None
self.stream = None
def setup_audio(self):
"""设置音频设备"""
try:
self.audio = pyaudio.PyAudio()
self.stream = self.audio.open(
format=self.FORMAT,
channels=self.CHANNELS,
rate=self.RATE,
input=True,
frames_per_buffer=self.CHUNK_SIZE
)
print("✅ 音频设备初始化成功")
return True
except Exception as e:
print(f"❌ 音频设备初始化失败: {e}")
return False
def calculate_zcr(self, audio_data):
"""计算零交叉率"""
if len(audio_data) == 0:
return 0
audio_array = np.frombuffer(audio_data, dtype=np.int16)
zero_crossings = np.sum(np.diff(np.sign(audio_array)) != 0)
zcr = zero_crossings / len(audio_array) * self.RATE
return zcr
def test_detection(self):
"""测试语音检测"""
print("🎙️ 开始语音检测测试")
print("=" * 50)
# 环境校准阶段
print("🔍 第一阶段:环境噪音校准 (10秒)")
print("请保持安静,不要说话...")
calibration_samples = []
start_time = time.time()
try:
while time.time() - start_time < 10:
data = self.stream.read(self.CHUNK_SIZE, exception_on_overflow=False)
if len(data) > 0:
zcr = self.calculate_zcr(data)
calibration_samples.append(zcr)
# 显示进度
progress = (time.time() - start_time) / 10 * 100
print(f"\r校准进度: {progress:.1f}%", end='', flush=True)
time.sleep(0.01)
print("\n✅ 环境校准完成")
# 计算统计数据
if calibration_samples:
avg_zcr = np.mean(calibration_samples)
std_zcr = np.std(calibration_samples)
min_zcr = min(calibration_samples)
max_zcr = max(calibration_samples)
print(f"📊 环境噪音统计:")
print(f" 平均ZCR: {avg_zcr:.0f}")
print(f" 标准差: {std_zcr:.0f}")
print(f" 最小值: {min_zcr:.0f}")
print(f" 最大值: {max_zcr:.0f}")
# 建议的检测阈值
suggested_min = max(2400, avg_zcr + 2 * std_zcr)
suggested_max = min(12000, avg_zcr + 6 * std_zcr)
print(f"\n🎯 建议的语音检测阈值:")
print(f" 最小阈值: {suggested_min:.0f}")
print(f" 最大阈值: {suggested_max:.0f}")
# 测试检测
print(f"\n🎙️ 第二阶段:语音检测测试 (20秒)")
print("现在请说话,测试语音检测...")
voice_threshold = suggested_min
silence_threshold = suggested_max
consecutive_voice = 0
voice_detected = False
test_start = time.time()
while time.time() - test_start < 20:
data = self.stream.read(self.CHUNK_SIZE, exception_on_overflow=False)
if len(data) > 0:
zcr = self.calculate_zcr(data)
# 简单的语音检测
is_voice = voice_threshold < zcr < silence_threshold
if is_voice:
consecutive_voice += 1
if consecutive_voice >= 5 and not voice_detected:
voice_detected = True
self.voice_count += 1
print(f"\n🎤 检测到语音 #{self.voice_count}! ZCR: {zcr:.0f}")
else:
consecutive_voice = 0
if voice_detected:
voice_detected = False
print(f" 语音结束,持续时间: {time.time() - last_voice_time:.1f}")
if voice_detected:
last_voice_time = time.time()
# 实时显示ZCR值
status = "🎤" if voice_detected else "🔇"
print(f"\r{status} ZCR: {zcr:.0f} | 阈值: {voice_threshold:.0f}-{silence_threshold:.0f} | "
f"连续语音: {consecutive_voice}/5", end='', flush=True)
time.sleep(0.01)
print(f"\n\n✅ 测试完成!共检测到 {self.voice_count} 次语音")
except KeyboardInterrupt:
print("\n🛑 测试被用户中断")
except Exception as e:
print(f"\n❌ 测试过程中出错: {e}")
def cleanup(self):
"""清理资源"""
if self.stream:
try:
self.stream.stop_stream()
self.stream.close()
except:
pass
if self.audio:
try:
self.audio.terminate()
except:
pass
def run_test(self):
"""运行完整测试"""
print("🚀 语音检测测试工具")
print("=" * 60)
if not self.setup_audio():
print("❌ 无法初始化音频设备,测试终止")
return
try:
self.test_detection()
finally:
self.cleanup()
print("\n👋 测试结束")
def main():
"""主函数"""
tester = VoiceDetectionTester()
tester.run_test()
if __name__ == "__main__":
main()

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@ -1,840 +0,0 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
语音交互聊天系统 - 集成豆包AI
基于能量检测的录音 + 豆包语音识别 + TTS回复
"""
import sys
import os
import time
import threading
import asyncio
import subprocess
import wave
import struct
import json
import gzip
import uuid
from typing import Dict, Any, Optional
import pyaudio
import numpy as np
import websockets
# 豆包协议常量
PROTOCOL_VERSION = 0b0001
CLIENT_FULL_REQUEST = 0b0001
CLIENT_AUDIO_ONLY_REQUEST = 0b0010
SERVER_FULL_RESPONSE = 0b1001
SERVER_ACK = 0b1011
SERVER_ERROR_RESPONSE = 0b1111
NO_SEQUENCE = 0b0000
MSG_WITH_EVENT = 0b0100
NO_SERIALIZATION = 0b0000
JSON = 0b0001
GZIP = 0b0001
class DoubaoClient:
"""豆包音频处理客户端"""
def __init__(self):
self.base_url = "wss://openspeech.bytedance.com/api/v3/realtime/dialogue"
self.app_id = "8718217928"
self.access_key = "ynJMX-5ix1FsJvswC9KTNlGUdubcchqc"
self.app_key = "PlgvMymc7f3tQnJ6"
self.resource_id = "volc.speech.dialog"
self.session_id = str(uuid.uuid4())
self.ws = None
self.log_id = ""
def get_headers(self) -> Dict[str, str]:
"""获取请求头"""
return {
"X-Api-App-ID": self.app_id,
"X-Api-Access-Key": self.access_key,
"X-Api-Resource-Id": self.resource_id,
"X-Api-App-Key": self.app_key,
"X-Api-Connect-Id": str(uuid.uuid4()),
}
def generate_header(self, message_type=CLIENT_FULL_REQUEST,
message_type_specific_flags=MSG_WITH_EVENT,
serial_method=JSON, compression_type=GZIP) -> bytes:
"""生成协议头"""
header = bytearray()
header.append((PROTOCOL_VERSION << 4) | 1) # version + header_size
header.append((message_type << 4) | message_type_specific_flags)
header.append((serial_method << 4) | compression_type)
header.append(0x00) # reserved
return bytes(header)
async def connect(self) -> None:
"""建立WebSocket连接"""
print(f"🔗 连接豆包服务器...")
try:
self.ws = await websockets.connect(
self.base_url,
additional_headers=self.get_headers(),
ping_interval=None
)
# 获取log_id
if hasattr(self.ws, 'response_headers'):
self.log_id = self.ws.response_headers.get("X-Tt-Logid")
elif hasattr(self.ws, 'headers'):
self.log_id = self.ws.headers.get("X-Tt-Logid")
print(f"✅ 连接成功, log_id: {self.log_id}")
# 发送StartConnection请求
await self._send_start_connection()
# 发送StartSession请求
await self._send_start_session()
except Exception as e:
print(f"❌ 连接失败: {e}")
raise
def parse_response(self, response):
"""解析响应"""
if len(response) < 4:
return None
protocol_version = response[0] >> 4
header_size = response[0] & 0x0f
message_type = response[1] >> 4
flags = response[1] & 0x0f
payload_start = header_size * 4
payload = response[payload_start:]
result = {
'protocol_version': protocol_version,
'header_size': header_size,
'message_type': message_type,
'flags': flags,
'payload': payload,
'payload_size': len(payload)
}
# 解析payload
if len(payload) >= 4:
result['event'] = int.from_bytes(payload[:4], 'big')
if len(payload) >= 8:
session_id_len = int.from_bytes(payload[4:8], 'big')
if len(payload) >= 8 + session_id_len:
result['session_id'] = payload[8:8+session_id_len].decode()
if len(payload) >= 12 + session_id_len:
data_size = int.from_bytes(payload[8+session_id_len:12+session_id_len], 'big')
result['data_size'] = data_size
result['data'] = payload[12+session_id_len:12+session_id_len+data_size]
# 尝试解析JSON数据
try:
result['json_data'] = json.loads(result['data'].decode('utf-8'))
except:
pass
return result
async def _send_start_connection(self) -> None:
"""发送StartConnection请求"""
request = bytearray(self.generate_header())
request.extend(int(1).to_bytes(4, 'big'))
payload_bytes = b"{}"
payload_bytes = gzip.compress(payload_bytes)
request.extend(len(payload_bytes).to_bytes(4, 'big'))
request.extend(payload_bytes)
await self.ws.send(request)
response = await self.ws.recv()
async def _send_start_session(self) -> None:
"""发送StartSession请求"""
session_config = {
"asr": {"extra": {"end_smooth_window_ms": 1500}},
"tts": {
"speaker": "zh_female_vv_jupiter_bigtts",
"audio_config": {"channel": 1, "format": "pcm", "sample_rate": 24000}
},
"dialog": {
"bot_name": "豆包",
"system_role": "你使用活泼灵动的女声,性格开朗,热爱生活。",
"speaking_style": "你的说话风格简洁明了,语速适中,语调自然。",
"location": {"city": "北京"},
"extra": {
"strict_audit": False,
"audit_response": "支持客户自定义安全审核回复话术。",
"recv_timeout": 30,
"input_mod": "audio",
},
},
}
request = bytearray(self.generate_header())
request.extend(int(100).to_bytes(4, 'big'))
request.extend(len(self.session_id).to_bytes(4, 'big'))
request.extend(self.session_id.encode())
payload_bytes = json.dumps(session_config).encode()
payload_bytes = gzip.compress(payload_bytes)
request.extend(len(payload_bytes).to_bytes(4, 'big'))
request.extend(payload_bytes)
await self.ws.send(request)
response = await self.ws.recv()
await asyncio.sleep(1.0)
async def process_audio(self, audio_data: bytes) -> tuple[str, bytes]:
"""处理音频并返回(识别文本, TTS音频)"""
try:
# 发送音频数据 - 使用与doubao_simple.py相同的格式
task_request = bytearray(
self.generate_header(message_type=CLIENT_AUDIO_ONLY_REQUEST,
serial_method=NO_SERIALIZATION))
task_request.extend(int(200).to_bytes(4, 'big'))
task_request.extend(len(self.session_id).to_bytes(4, 'big'))
task_request.extend(self.session_id.encode())
payload_bytes = gzip.compress(audio_data)
task_request.extend(len(payload_bytes).to_bytes(4, 'big'))
task_request.extend(payload_bytes)
await self.ws.send(task_request)
print("📤 音频数据已发送")
recognized_text = ""
tts_audio = b""
response_count = 0
# 接收响应 - 使用与doubao_simple.py相同的解析逻辑
audio_chunks = []
max_responses = 30
while response_count < max_responses:
try:
response = await asyncio.wait_for(self.ws.recv(), timeout=30.0)
response_count += 1
parsed = self.parse_response(response)
if not parsed:
continue
print(f"📥 响应 {response_count}: message_type={parsed['message_type']}, event={parsed.get('event', 'N/A')}, size={parsed['payload_size']}")
# 处理不同类型的响应
if parsed['message_type'] == 11: # SERVER_ACK - 可能包含音频
if 'data' in parsed and parsed['data_size'] > 0:
audio_chunks.append(parsed['data'])
print(f"收集到音频块: {parsed['data_size']} 字节")
elif parsed['message_type'] == 9: # SERVER_FULL_RESPONSE
event = parsed.get('event', 0)
if event == 450: # ASR开始
print("🎤 ASR处理开始")
elif event == 451: # ASR结果
if 'json_data' in parsed and 'results' in parsed['json_data']:
text = parsed['json_data']['results'][0].get('text', '')
recognized_text = text
print(f"🧠 识别结果: {text}")
elif event == 459: # ASR结束
print("✅ ASR处理结束")
elif event == 350: # TTS开始
print("🎵 TTS生成开始")
elif event == 359: # TTS结束
print("✅ TTS生成结束")
break
elif event == 550: # TTS音频数据
if 'data' in parsed and parsed['data_size'] > 0:
# 检查是否是JSON音频元数据还是实际音频数据
try:
json.loads(parsed['data'].decode('utf-8'))
print("收到TTS音频元数据")
except:
# 不是JSON可能是音频数据
audio_chunks.append(parsed['data'])
print(f"收集到TTS音频块: {parsed['data_size']} 字节")
except asyncio.TimeoutError:
print(f"⏰ 等待响应 {response_count + 1} 超时")
break
except websockets.exceptions.ConnectionClosed:
print("🔌 连接已关闭")
break
print(f"共收到 {response_count} 个响应,收集到 {len(audio_chunks)} 个音频块")
# 合并音频数据
if audio_chunks:
tts_audio = b''.join(audio_chunks)
print(f"合并后的音频数据: {len(tts_audio)} 字节")
# 转换TTS音频格式32位浮点 -> 16位整数
if tts_audio:
# 检查是否是GZIP压缩数据
try:
decompressed = gzip.decompress(tts_audio)
print(f"解压缩后音频数据: {len(decompressed)} 字节")
audio_to_write = decompressed
except:
print("音频数据不是GZIP压缩格式直接使用原始数据")
audio_to_write = tts_audio
# 检查音频数据长度是否是4的倍数32位浮点
if len(audio_to_write) % 4 != 0:
print(f"警告:音频数据长度 {len(audio_to_write)} 不是4的倍数截断到最近的倍数")
audio_to_write = audio_to_write[:len(audio_to_write) // 4 * 4]
# 将32位浮点转换为16位整数
float_count = len(audio_to_write) // 4
int16_data = bytearray(float_count * 2)
for i in range(float_count):
# 读取32位浮点数小端序
float_value = struct.unpack('<f', audio_to_write[i*4:i*4+4])[0]
# 将浮点数限制在[-1.0, 1.0]范围内
float_value = max(-1.0, min(1.0, float_value))
# 转换为16位整数
int16_value = int(float_value * 32767)
# 写入16位整数小端序
int16_data[i*2:i*2+2] = struct.pack('<h', int16_value)
tts_audio = bytes(int16_data)
print(f"✅ 音频转换完成: {len(tts_audio)} 字节")
return recognized_text, tts_audio
except Exception as e:
print(f"❌ 处理失败: {e}")
import traceback
traceback.print_exc()
return "", b""
async def send_silence_data(self, duration_ms=100) -> None:
"""发送静音数据保持连接活跃"""
try:
# 生成静音音频数据
samples = int(16000 * duration_ms / 1000) # 16kHz采样率
silence_data = bytes(samples * 2) # 16位PCM
# 发送静音数据
task_request = bytearray(
self.generate_header(message_type=CLIENT_AUDIO_ONLY_REQUEST,
serial_method=NO_SERIALIZATION))
task_request.extend(int(200).to_bytes(4, 'big'))
task_request.extend(len(self.session_id).to_bytes(4, 'big'))
task_request.extend(self.session_id.encode())
payload_bytes = gzip.compress(silence_data)
task_request.extend(len(payload_bytes).to_bytes(4, 'big'))
task_request.extend(payload_bytes)
await self.ws.send(task_request)
print("💓 发送心跳数据保持连接")
# 简单处理响应(不等待完整响应)
try:
response = await asyncio.wait_for(self.ws.recv(), timeout=5.0)
# 只确认收到响应,不处理内容
except asyncio.TimeoutError:
print("⚠️ 心跳响应超时")
except websockets.exceptions.ConnectionClosed:
print("❌ 心跳时连接已关闭")
raise
except Exception as e:
print(f"❌ 发送心跳数据失败: {e}")
async def close(self) -> None:
"""关闭连接"""
if self.ws:
try:
await self.ws.close()
except:
pass
print("🔌 连接已关闭")
class VoiceChatRecorder:
"""语音聊天录音系统"""
def __init__(self, enable_ai_chat=True):
# 音频参数
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
self.RATE = 16000
self.CHUNK_SIZE = 1024
# 能量检测参数
self.energy_threshold = 500
self.silence_threshold = 2.0
self.min_recording_time = 1.0
self.max_recording_time = 20.0
# 状态变量
self.audio = None
self.stream = None
self.running = False
self.recording = False
self.recorded_frames = []
self.recording_start_time = None
self.last_sound_time = None
self.energy_history = []
self.zcr_history = []
# AI聊天功能
self.enable_ai_chat = enable_ai_chat
self.doubao_client = None
self.is_processing_ai = False
self.heartbeat_thread = None
self.last_heartbeat_time = time.time()
self.heartbeat_interval = 10.0 # 每10秒发送一次心跳
# 预录音缓冲区
self.pre_record_buffer = []
self.pre_record_max_frames = int(2.0 * self.RATE / self.CHUNK_SIZE)
# 播放状态
self.is_playing = False
# ZCR检测参数
self.consecutive_low_zcr_count = 0
self.low_zcr_threshold_count = 15
self.voice_activity_history = []
self._setup_audio()
def _setup_audio(self):
"""设置音频设备"""
try:
self.audio = pyaudio.PyAudio()
self.stream = self.audio.open(
format=self.FORMAT,
channels=self.CHANNELS,
rate=self.RATE,
input=True,
frames_per_buffer=self.CHUNK_SIZE
)
print("✅ 音频设备初始化成功")
except Exception as e:
print(f"❌ 音频设备初始化失败: {e}")
def generate_silence_audio(self, duration_ms=100):
"""生成静音音频数据"""
# 生成指定时长的静音音频16位PCM值为0
samples = int(self.RATE * duration_ms / 1000)
silence_data = bytes(samples * 2) # 16位 = 2字节每样本
return silence_data
def calculate_energy(self, audio_data):
"""计算音频能量"""
if len(audio_data) == 0:
return 0
audio_array = np.frombuffer(audio_data, dtype=np.int16)
rms = np.sqrt(np.mean(audio_array ** 2))
if not self.recording:
self.energy_history.append(rms)
if len(self.energy_history) > 50:
self.energy_history.pop(0)
return rms
def calculate_zero_crossing_rate(self, audio_data):
"""计算零交叉率"""
if len(audio_data) == 0:
return 0
audio_array = np.frombuffer(audio_data, dtype=np.int16)
zero_crossings = np.sum(np.diff(np.sign(audio_array)) != 0)
zcr = zero_crossings / len(audio_array) * self.RATE
self.zcr_history.append(zcr)
if len(self.zcr_history) > 30:
self.zcr_history.pop(0)
return zcr
def is_voice_active(self, energy, zcr):
"""使用ZCR进行语音活动检测"""
# 16000Hz采样率下的语音ZCR范围
zcr_condition = 2400 < zcr < 12000
return zcr_condition
def save_recording(self, audio_data, filename=None):
"""保存录音"""
if filename is None:
timestamp = time.strftime("%Y%m%d_%H%M%S")
filename = f"recording_{timestamp}.wav"
try:
with wave.open(filename, 'wb') as wf:
wf.setnchannels(self.CHANNELS)
wf.setsampwidth(self.audio.get_sample_size(self.FORMAT))
wf.setframerate(self.RATE)
wf.writeframes(audio_data)
print(f"✅ 录音已保存: {filename}")
return True, filename
except Exception as e:
print(f"❌ 保存录音失败: {e}")
return False, None
def play_audio(self, filename):
"""播放音频文件"""
try:
# 停止当前录音
if self.recording:
self.recording = False
self.recorded_frames = []
# 关闭输入流
if self.stream:
self.stream.stop_stream()
self.stream.close()
self.stream = None
self.is_playing = True
time.sleep(0.2)
# 使用系统播放器
print(f"🔊 播放: {filename}")
subprocess.run(['aplay', filename], check=True)
print("✅ 播放完成")
except Exception as e:
print(f"❌ 播放失败: {e}")
finally:
self.is_playing = False
time.sleep(0.2)
self._setup_audio()
def update_pre_record_buffer(self, audio_data):
"""更新预录音缓冲区"""
self.pre_record_buffer.append(audio_data)
if len(self.pre_record_buffer) > self.pre_record_max_frames:
self.pre_record_buffer.pop(0)
def start_recording(self):
"""开始录音"""
print("🎙️ 检测到声音,开始录音...")
self.recording = True
self.recorded_frames = []
self.recorded_frames.extend(self.pre_record_buffer)
self.pre_record_buffer = []
self.recording_start_time = time.time()
self.last_sound_time = time.time()
self.consecutive_low_zcr_count = 0
def stop_recording(self):
"""停止录音"""
if len(self.recorded_frames) > 0:
audio_data = b''.join(self.recorded_frames)
duration = len(audio_data) / (self.RATE * 2)
print(f"📝 录音完成,时长: {duration:.2f}")
if self.enable_ai_chat:
# AI聊天模式
self.process_with_ai(audio_data)
else:
# 普通录音模式
success, filename = self.save_recording(audio_data)
if success and filename:
print("=" * 50)
print("🔊 播放刚才录制的音频...")
self.play_audio(filename)
print("=" * 50)
self.recording = False
self.recorded_frames = []
self.recording_start_time = None
self.last_sound_time = None
def process_with_ai(self, audio_data):
"""使用AI处理录音"""
if self.is_processing_ai:
print("⏳ AI正在处理中请稍候...")
return
self.is_processing_ai = True
# 在新线程中处理AI
ai_thread = threading.Thread(target=self._ai_processing_thread, args=(audio_data,))
ai_thread.daemon = True
ai_thread.start()
def _heartbeat_thread(self):
"""心跳线程 - 定期发送静音数据保持连接活跃"""
while self.running and self.doubao_client and self.doubao_client.ws:
current_time = time.time()
if current_time - self.last_heartbeat_time >= self.heartbeat_interval:
try:
# 异步发送心跳数据
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
try:
loop.run_until_complete(self.doubao_client.send_silence_data())
self.last_heartbeat_time = current_time
except Exception as e:
print(f"❌ 心跳失败: {e}")
# 如果心跳失败,可能需要重新连接
break
finally:
loop.close()
except Exception as e:
print(f"❌ 心跳线程异常: {e}")
break
# 睡眠一段时间
time.sleep(1.0)
print("📡 心跳线程结束")
def _ai_processing_thread(self, audio_data):
"""AI处理线程"""
try:
print("🤖 开始AI处理...")
print("🧠 正在进行语音识别...")
# 异步处理
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
try:
# 连接豆包
self.doubao_client = DoubaoClient()
loop.run_until_complete(self.doubao_client.connect())
# 启动心跳线程
self.last_heartbeat_time = time.time()
self.heartbeat_thread = threading.Thread(target=self._heartbeat_thread)
self.heartbeat_thread.daemon = True
self.heartbeat_thread.start()
print("💓 心跳线程已启动")
# 语音识别和TTS回复
recognized_text, tts_audio = loop.run_until_complete(
self.doubao_client.process_audio(audio_data)
)
if recognized_text:
print(f"🗣️ 你说: {recognized_text}")
if tts_audio:
# 保存TTS音频
tts_filename = "ai_response.wav"
with wave.open(tts_filename, 'wb') as wav_file:
wav_file.setnchannels(1)
wav_file.setsampwidth(2)
wav_file.setframerate(24000)
wav_file.writeframes(tts_audio)
print("🎵 AI回复生成完成")
print("=" * 50)
print("🔊 播放AI回复...")
self.play_audio(tts_filename)
print("=" * 50)
else:
print("❌ 未收到AI回复")
# 等待一段时间再关闭连接,以便心跳继续工作
print("⏳ 等待5秒后关闭连接...")
time.sleep(5)
except Exception as e:
print(f"❌ AI处理失败: {e}")
finally:
# 停止心跳线程
if self.heartbeat_thread and self.heartbeat_thread.is_alive():
print("🛑 停止心跳线程")
self.heartbeat_thread = None
# 关闭连接
if self.doubao_client:
loop.run_until_complete(self.doubao_client.close())
loop.close()
except Exception as e:
print(f"❌ AI处理线程失败: {e}")
finally:
self.is_processing_ai = False
def run(self):
"""运行语音聊天系统"""
if not self.stream:
print("❌ 音频设备未初始化")
return
self.running = True
if self.enable_ai_chat:
print("🤖 语音聊天AI助手")
print("=" * 50)
print("🎯 功能特点:")
print("- 🎙️ 智能语音检测")
print("- 🧠 豆包AI语音识别")
print("- 🗣️ AI智能回复")
print("- 🔊 TTS语音播放")
print("- 🔄 实时对话")
print("=" * 50)
print("📖 使用说明:")
print("- 说话自动录音")
print("- 静音2秒结束录音")
print("- AI自动识别并回复")
print("- 按 Ctrl+C 退出")
print("=" * 50)
else:
print("🎙️ 智能录音系统")
print("=" * 50)
print("📖 使用说明:")
print("- 说话自动录音")
print("- 静音2秒结束录音")
print("- 录音完成后自动播放")
print("- 按 Ctrl+C 退出")
print("=" * 50)
try:
while self.running:
# 如果正在播放AI回复跳过音频处理
if self.is_playing or self.is_processing_ai:
status = "🤖 AI处理中..."
print(f"\r{status}", end='', flush=True)
time.sleep(0.1)
continue
# 读取音频数据
data = self.stream.read(self.CHUNK_SIZE, exception_on_overflow=False)
if len(data) == 0:
continue
# 计算能量和ZCR
energy = self.calculate_energy(data)
zcr = self.calculate_zero_crossing_rate(data)
if self.recording:
# 录音模式
self.recorded_frames.append(data)
recording_duration = time.time() - self.recording_start_time
# 检测语音活动
if self.is_voice_active(energy, zcr):
self.last_sound_time = time.time()
self.consecutive_low_zcr_count = 0
else:
self.consecutive_low_zcr_count += 1
# 检查是否应该结束录音
should_stop = False
# ZCR静音检测
if self.consecutive_low_zcr_count >= self.low_zcr_threshold_count:
should_stop = True
# 时间静音检测
if not should_stop and time.time() - self.last_sound_time > self.silence_threshold:
should_stop = True
# 执行停止录音
if should_stop and recording_duration >= self.min_recording_time:
print(f"\n🔇 检测到静音,结束录音")
self.stop_recording()
# 检查最大录音时间
if recording_duration > self.max_recording_time:
print(f"\n⏰ 达到最大录音时间")
self.stop_recording()
# 显示录音状态
is_voice = self.is_voice_active(energy, zcr)
zcr_count = f"{self.consecutive_low_zcr_count}/{self.low_zcr_threshold_count}"
status = f"录音中... {recording_duration:.1f}s | ZCR: {zcr:.0f} | 语音: {is_voice} | 静音计数: {zcr_count}"
print(f"\r{status}", end='', flush=True)
else:
# 监听模式
self.update_pre_record_buffer(data)
if self.is_voice_active(energy, zcr):
# 检测到声音,开始录音
self.start_recording()
else:
# 显示监听状态
is_voice = self.is_voice_active(energy, zcr)
buffer_usage = len(self.pre_record_buffer) / self.pre_record_max_frames * 100
status = f"监听中... ZCR: {zcr:.0f} | 语音: {is_voice} | 缓冲: {buffer_usage:.0f}%"
print(f"\r{status}", end='', flush=True)
time.sleep(0.01)
except KeyboardInterrupt:
print("\n👋 退出")
except Exception as e:
print(f"❌ 错误: {e}")
finally:
self.stop()
def stop(self):
"""停止系统"""
self.running = False
# 停止心跳线程
if self.heartbeat_thread and self.heartbeat_thread.is_alive():
print("🛑 停止心跳线程")
self.heartbeat_thread = None
if self.recording:
self.stop_recording()
if self.stream:
self.stream.stop_stream()
self.stream.close()
if self.audio:
self.audio.terminate()
# 关闭AI连接
if self.doubao_client and self.doubao_client.ws:
try:
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
loop.run_until_complete(self.doubao_client.close())
loop.close()
except:
pass
def main():
"""主函数"""
import argparse
parser = argparse.ArgumentParser(description='语音聊天AI助手')
parser.add_argument('--no-ai', action='store_true', help='禁用AI功能仅录音')
args = parser.parse_args()
enable_ai = not args.no_ai
if enable_ai:
print("🚀 语音聊天AI助手")
else:
print("🚀 智能录音系统")
print("=" * 50)
# 创建语音聊天系统
recorder = VoiceChatRecorder(enable_ai_chat=enable_ai)
print("✅ 系统初始化成功")
print("=" * 50)
# 开始运行
recorder.run()
if __name__ == "__main__":
main()

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zcr_monitor.py Normal file
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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
实时ZCR监控工具
用于观察实际的ZCR值和测试语音检测
"""
import threading
import time
import numpy as np
import pyaudio
class ZCRMonitor:
"""ZCR实时监控器"""
def __init__(self):
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
self.RATE = 16000
self.CHUNK_SIZE = 1024
# 监控参数
self.running = False
self.zcr_history = []
self.max_history = 100
# 音频设备
self.audio = None
self.stream = None
# 检测阈值匹配recorder.py的设置
self.zcr_min = 2400
self.zcr_max = 12000
def setup_audio(self):
"""设置音频设备"""
try:
self.audio = pyaudio.PyAudio()
self.stream = self.audio.open(
format=self.FORMAT,
channels=self.CHANNELS,
rate=self.RATE,
input=True,
frames_per_buffer=self.CHUNK_SIZE
)
return True
except Exception as e:
print(f"❌ 音频设备初始化失败: {e}")
return False
def calculate_zcr(self, audio_data):
"""计算零交叉率"""
if len(audio_data) == 0:
return 0
audio_array = np.frombuffer(audio_data, dtype=np.int16)
zero_crossings = np.sum(np.diff(np.sign(audio_array)) != 0)
zcr = zero_crossings / len(audio_array) * self.RATE
return zcr
def is_voice(self, zcr):
"""简单的语音检测"""
return self.zcr_min < zcr < self.zcr_max
def monitor_callback(self, in_data, frame_count, time_info, status):
"""音频回调函数"""
zcr = self.calculate_zcr(in_data)
# 更新历史
self.zcr_history.append(zcr)
if len(self.zcr_history) > self.max_history:
self.zcr_history.pop(0)
# 计算统计信息
if len(self.zcr_history) > 10:
avg_zcr = np.mean(self.zcr_history[-10:]) # 最近10个值的平均
std_zcr = np.std(self.zcr_history[-10:])
else:
avg_zcr = zcr
std_zcr = 0
# 判断是否为语音
voice_detected = self.is_voice(zcr)
# 实时显示
status = "🎤" if voice_detected else "🔇"
color = "\033[92m" if voice_detected else "\033[90m" # 绿色或灰色
reset = "\033[0m"
# 显示信息
info = (f"{color}{status} ZCR: {zcr:.0f} | "
f"阈值: {self.zcr_min}-{self.zcr_max} | "
f"平均: {avg_zcr:.0f}±{std_zcr:.0f}{reset}")
print(f"\r{info}", end='', flush=True)
return (in_data, pyaudio.paContinue)
def start_monitoring(self):
"""开始监控"""
print("🎙️ ZCR实时监控工具")
print("=" * 50)
print("📊 当前检测阈值:")
print(f" ZCR范围: {self.zcr_min} - {self.zcr_max}")
print("💡 请说话测试语音检测...")
print("🛑 按 Ctrl+C 停止监控")
print("=" * 50)
try:
# 使用回调模式
self.stream = self.audio.open(
format=self.FORMAT,
channels=self.CHANNELS,
rate=self.RATE,
input=True,
frames_per_buffer=self.CHUNK_SIZE,
stream_callback=self.monitor_callback
)
self.stream.start_stream()
self.running = True
# 主循环
while self.running:
time.sleep(0.1)
except KeyboardInterrupt:
print("\n🛑 监控停止")
finally:
self.cleanup()
def show_statistics(self):
"""显示统计信息"""
if not self.zcr_history:
return
print("\n📊 ZCR统计信息:")
print(f" 样本数量: {len(self.zcr_history)}")
print(f" 最小值: {min(self.zcr_history):.0f}")
print(f" 最大值: {max(self.zcr_history):.0f}")
print(f" 平均值: {np.mean(self.zcr_history):.0f}")
print(f" 标准差: {np.std(self.zcr_history):.0f}")
# 分析语音检测
voice_count = sum(1 for zcr in self.zcr_history if self.is_voice(zcr))
voice_percentage = voice_count / len(self.zcr_history) * 100
print(f" 语音检测: {voice_count}/{len(self.zcr_history)} ({voice_percentage:.1f}%)")
# 建议新的阈值
avg_zcr = np.mean(self.zcr_history)
std_zcr = np.std(self.zcr_history)
suggested_min = max(800, avg_zcr + std_zcr)
suggested_max = min(8000, avg_zcr + 4 * std_zcr)
print(f"\n🎯 建议的检测阈值:")
print(f" 最小值: {suggested_min:.0f}")
print(f" 最大值: {suggested_max:.0f}")
def cleanup(self):
"""清理资源"""
self.running = False
if self.stream:
try:
self.stream.stop_stream()
self.stream.close()
except:
pass
if self.audio:
try:
self.audio.terminate()
except:
pass
# 显示最终统计
self.show_statistics()
def main():
"""主函数"""
monitor = ZCRMonitor()
if not monitor.setup_audio():
print("❌ 无法初始化音频设备")
return
try:
monitor.start_monitoring()
except Exception as e:
print(f"❌ 监控过程中出错: {e}")
finally:
monitor.cleanup()
if __name__ == "__main__":
main()