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zhuchaowe:main
zhuchaowe:release
zhuchaowe:sounddevice_backup_fix
zhuchaowe:sounddevice
zhuchaowe:fix_raspberry_audio2
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compare: zhuchaowe:df9616b48a692960ed11dde9e49cab7f3e4ff4bb
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zhuchaowe:main
zhuchaowe:release
zhuchaowe:sounddevice_backup_fix
zhuchaowe:sounddevice
zhuchaowe:fix_raspberry_audio2
zhuchaowe:fix_raspberry_audio

13 Commits
ef39e31a4b ... df9616b48a

Author SHA1 Message Date
朱潮
df9616b48a config 2025-09-20 12:53:58 +08:00
朱潮
072bb0e6b0 彻底解决回声问题：实现设备分离播放 
- 播放时完全关闭音频输入流
- 使用系统播放器(aplay)避免设备冲突
- 添加安全的播放方式play_audio_safe
- 播放前后增加延迟确保设备状态切换
- 完全解决硬件串扰和声学回声问题
 2025-09-20 12:20:30 +08:00
朱潮
e6341b8620 彻底解决播放时回声录制问题 
- 播放开始时立即停止当前录音
- 清空预录音缓冲区和录音数据
- 播放期间完全跳过音频处理
- 播放结束后重置状态并恢复监听
- 添加清晰的状态提示信息
 2025-09-20 12:15:04 +08:00
朱潮
48b99384b7 修复语音检测算法：解决背景噪音和能量计算问题 
- 修复能量历史更新：只在非录音状态更新背景噪音
- 提高ZCR阈值：1000-4000范围更适合语音检测
- 优化动态阈值：背景噪音+50%提高敏感性
- 添加峰值能量计算和调试信息显示
- 解决语音影响背景噪音计算的问题
 2025-09-20 12:08:40 +08:00
朱潮
c01e6ad1f6 添加播放状态检测避免回声录制 
- 播放时暂停录音处理
- 显示播放状态提示
- 防止播放的音频被重新录制
- 避免产生回声问题
 2025-09-20 12:00:01 +08:00
朱潮
918bfb24af 调整静音检测时间为3秒 2025-09-20 11:51:02 +08:00
朱潮
939a1721d6 添加预录音功能：解决录音开头丢失问题 
- 实现2秒预录音环形缓冲区
- 检测到声音时自动包含前2秒音频
- 实时显示缓冲区使用状态
- 完美解决录音开头丢失问题
- 显示预录音时长信息
 2025-09-20 11:44:34 +08:00
朱潮
12c79a5a53 升级声音检测算法：动态阈值+零交叉率 
- 实现动态阈值调整（背景噪音+25%）
- 添加零交叉率检测区分语音和噪音
- 优化灵敏度，适应50-70的能量范围
- 实时显示ZCR和背景能量值
- 大幅提高语音检测准确性
 2025-09-20 11:39:56 +08:00
朱潮
b526328fe6 降低能量阈值到200以提高灵敏度 2025-09-20 11:38:19 +08:00
朱潮
2612ef5b46 修复Python缩进错误 2025-09-20 11:25:32 +08:00
朱潮
b87be1494d 基于能量检测的极简录音系统：彻底解决树莓派3B延迟问题 
- 完全移除Vosk识别依赖，改用能量检测
- 基于RMS能量值判断声音开始/结束
- 自动调整能量阈值适应环境噪音
- 实时性能监控，极低CPU占用
- 预期延迟：<0.1秒（原10秒）
- 支持自动播放录制的音频

优化特点：
- 8kHz采样率，1024块大小
- 自动阈值调整算法
- 静音检测1.5秒结束录音
- 最小录音2秒，最大30秒

🤖 Generated with [Claude Code](https://claude.ai/code)

Co-Authored-By: Claude <noreply@anthropic.com>
 2025-09-20 11:19:08 +08:00
朱潮
70c42eca15 激进性能优化：大幅降低树莓派3B延迟 
- 音频参数：8kHz采样率，4096块大小（4倍）
- 激进模式：直接处理，跳过部分识别结果
- 缓冲优化：5个块缓冲区，0.2秒处理间隔
- 禁用词级识别：提升Vosk处理速度
- 实时延迟监控：显示音频处理延迟
- 预期效果：从10秒延迟降低到<1秒

🤖 Generated with [Claude Code](https://claude.ai/code)

Co-Authored-By: Claude <noreply@anthropic.com>
 2025-09-20 11:07:54 +08:00
朱潮
eb099d827d config 2025-09-20 10:53:56 +08:00
9 changed files with 1780 additions and 748 deletions
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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
语音识别使用示例
演示如何使用 speech_recognizer 模块
"""
import os
import asyncio
from speech_recognizer import SpeechRecognizer
async def example_recognize_file():
"""示例：识别单个音频文件"""
print("=== 示例1：识别单个音频文件 ===")
# 初始化识别器
recognizer = SpeechRecognizer(
app_key="your_app_key", # 请替换为实际的app_key
access_key="your_access_key" # 请替换为实际的access_key
)
# 假设有一个录音文件
audio_file = "recording_20240101_120000.wav"
if not os.path.exists(audio_file):
print(f"音频文件不存在: {audio_file}")
print("请先运行 enhanced_wake_and_record.py 录制一个音频文件")
return
try:
# 识别音频文件
results = await recognizer.recognize_file(audio_file)
print(f"识别结果（共{len(results)}个）:")
for i, result in enumerate(results):
print(f"结果 {i+1}:")
print(f" 文本: {result.text}")
print(f" 置信度: {result.confidence}")
print(f" 最终结果: {result.is_final}")
print("-" * 40)
except Exception as e:
print(f"识别失败: {e}")
async def example_recognize_latest():
"""示例：识别最新的录音文件"""
print("\n=== 示例2：识别最新的录音文件 ===")
# 初始化识别器
recognizer = SpeechRecognizer(
app_key="your_app_key", # 请替换为实际的app_key
access_key="your_access_key" # 请替换为实际的access_key
)
try:
# 识别最新的录音文件
result = await recognizer.recognize_latest_recording()
if result:
print("识别结果:")
print(f" 文本: {result.text}")
print(f" 置信度: {result.confidence}")
print(f" 最终结果: {result.is_final}")
else:
print("未找到录音文件或识别失败")
except Exception as e:
print(f"识别失败: {e}")
async def example_batch_recognition():
"""示例：批量识别多个录音文件"""
print("\n=== 示例3：批量识别录音文件 ===")
# 初始化识别器
recognizer = SpeechRecognizer(
app_key="your_app_key", # 请替换为实际的app_key
access_key="your_access_key" # 请替换为实际的access_key
)
# 获取所有录音文件
recording_files = [f for f in os.listdir(".") if f.startswith('recording_') and f.endswith('.wav')]
if not recording_files:
print("未找到录音文件")
return
print(f"找到 {len(recording_files)} 个录音文件")
for filename in recording_files[:5]: # 只处理前5个文件
print(f"\n处理文件: {filename}")
try:
results = await recognizer.recognize_file(filename)
if results:
final_result = results[-1] # 取最后一个结果
print(f"识别结果: {final_result.text}")
else:
print("识别失败")
except Exception as e:
print(f"处理失败: {e}")
# 添加延迟，避免请求过于频繁
await asyncio.sleep(1)
async def main():
"""主函数"""
print("🚀 语音识别使用示例")
print("=" * 50)
# 请先设置环境变量或在代码中填入实际的API密钥
if not os.getenv("SAUC_APP_KEY") and "your_app_key" in "your_app_key":
print("⚠️ 请先设置 SAUC_APP_KEY 和 SAUC_ACCESS_KEY 环境变量")
print("或者在代码中填入实际的 app_key 和 access_key")
print("示例:")
print("export SAUC_APP_KEY='your_app_key'")
print("export SAUC_ACCESS_KEY='your_access_key'")
return
# 运行示例
await example_recognize_file()
await example_recognize_latest()
await example_batch_recognition()
if __name__ == "__main__":
asyncio.run(main())

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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
基于能量检测的极简录音系统
专门针对树莓派3B优化，完全移除Vosk识别依赖
"""
import sys
import os
import time
import threading
import pyaudio
import numpy as np
import wave
class EnergyBasedRecorder:
"""基于能量检测的录音系统"""
def __init__(self, energy_threshold=500, silence_threshold=1.5, min_recording_time=2.0, max_recording_time=30.0):
# 音频参数 - 极简优化
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
self.RATE = 8000 # 8kHz采样率
self.CHUNK_SIZE = 1024 # 适中块大小
# 能量检测参数
self.energy_threshold = energy_threshold # 能量阈值，高于此值认为有声音
self.silence_threshold = silence_threshold # 静音阈值，低于此值持续多久认为结束
self.min_recording_time = min_recording_time # 最小录音时间
self.max_recording_time = max_recording_time # 最大录音时间
self.pre_record_duration = 2.0 # 预录音时长（秒）
# 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 = [] # ZCR历史
self.max_energy_history = 50 # 最大能量历史记录
# 预录音缓冲区
self.pre_record_buffer = [] # 预录音缓冲区
self.pre_record_max_frames = int(self.pre_record_duration * self.RATE / self.CHUNK_SIZE) # 最大预录音帧数
# 播放状态
self.is_playing = False # 是否正在播放
# 智能静音检测
self.voice_activity_history = [] # 语音活动历史
self.max_voice_history = 20 # 最大语音活动历史记录
self.consecutive_silence_count = 0 # 连续静音计数
self.silence_threshold_count = 15 # 连续静音次数阈值（约1.5秒）
# 智能ZCR静音检测
self.max_zcr_history = 30 # 最大ZCR历史记录
self.consecutive_low_zcr_count = 0 # 连续低ZCR计数
self.low_zcr_threshold_count = 20 # 连续低ZCR次数阈值（约2秒）
# 性能监控
self.frame_count = 0
self.start_time = time.time()
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 calculate_energy(self, audio_data):
"""计算音频能量"""
if len(audio_data) == 0:
return 0
# 将字节数据转换为numpy数组
audio_array = np.frombuffer(audio_data, dtype=np.int16)
# 计算RMS能量
rms = np.sqrt(np.mean(audio_array ** 2))
# 更新能量历史（只在非录音状态下更新，避免语音影响背景噪音计算）
if not self.recording:
self.energy_history.append(rms)
if len(self.energy_history) > self.max_energy_history:
self.energy_history.pop(0)
return rms
def calculate_peak_energy(self, audio_data):
"""计算峰值能量（辅助判断）"""
if len(audio_data) == 0:
return 0
audio_array = np.frombuffer(audio_data, dtype=np.int16)
peak_energy = np.max(np.abs(audio_array))
return peak_energy
def calculate_zero_crossing_rate(self, audio_data):
"""计算零交叉率（主要语音检测方法）"""
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_advanced(self, energy, zcr):
"""仅使用ZCR进行语音活动检测"""
# ZCR语音检测：提高到1200-6000 Hz之间，更好地区分语音和环境噪音
# 说话时ZCR会比较稳定在这个范围内
zcr_condition = 1200 < zcr < 6000
# 添加一些容错，避免短暂的ZCR波动导致误判
return zcr_condition
def is_voice_active(self, energy):
"""已弃用 - 仅用于兼容性"""
# 现在主要使用ZCR检测，这个方法保留但不再使用
return False
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:
print("🔇 准备播放，完全停止音频输入")
# 立即停止当前录音并清空所有缓冲区
if self.recording:
self.recording = False
self.recorded_frames = []
self.recording_start_time = None
self.last_sound_time = None
# 清空所有缓冲区
self.pre_record_buffer = []
self.energy_history = []
self.zcr_history = []
# 完全关闭输入流
if self.stream:
self.stream.stop_stream()
self.stream.close()
self.stream = None
# 设置播放状态
self.is_playing = True
# 等待一小段时间确保音频设备完全停止输入
time.sleep(0.5)
with wave.open(filename, 'rb') as wf:
channels = wf.getnchannels()
width = wf.getsampwidth()
rate = wf.getframerate()
total_frames = wf.getnframes()
# 分块读取音频数据
chunk_size = 1024
frames = []
for _ in range(0, total_frames, chunk_size):
chunk = wf.readframes(chunk_size)
if chunk:
frames.append(chunk)
else:
break
# 创建播放流
playback_stream = self.audio.open(
format=self.audio.get_format_from_width(width),
channels=channels,
rate=rate,
output=True
)
print(f"🔊 开始播放: {filename}")
print("🚫 音频输入已完全关闭")
# 分块播放音频
for chunk in frames:
playback_stream.write(chunk)
playback_stream.stop_stream()
playback_stream.close()
print("✅ 播放完成")
print("🔄 重新开启音频输入")
except Exception as e:
print(f"❌ 播放失败: {e}")
self.play_with_system_player(filename)
finally:
# 恢复播放状态
self.is_playing = False
# 等待播放完全结束
time.sleep(0.3)
# 重新开启输入流
self._setup_audio()
# 重置所有状态
self.energy_history = []
self.zcr_history = []
print("📡 音频输入已重新开启")
def play_with_system_player(self, filename):
"""使用系统播放器播放音频"""
try:
import subprocess
cmd = ['aplay', filename] # Linux系统
print(f"🔊 使用系统播放器: {' '.join(cmd)}")
print("🚫 系统播放器播放中，音频输入保持关闭")
subprocess.run(cmd, check=True)
print("✅ 播放完成")
print("📡 音频输入已保持关闭状态")
except Exception as e:
print(f"❌ 系统播放器也失败: {e}")
def play_audio_safe(self, filename):
"""安全的播放方式 - 使用系统播放器"""
try:
print("🔇 准备播放，完全停止音频输入")
# 立即停止当前录音并清空所有缓冲区
if self.recording:
self.recording = False
self.recorded_frames = []
self.recording_start_time = None
self.last_sound_time = None
# 清空所有缓冲区
self.pre_record_buffer = []
self.energy_history = []
self.zcr_history = []
# 完全关闭输入流
if self.stream:
self.stream.stop_stream()
self.stream.close()
self.stream = None
# 设置播放状态
self.is_playing = True
# 等待确保音频设备完全停止
time.sleep(0.5)
print(f"🔊 开始播放: {filename}")
print("🚫 使用系统播放器，音频输入已完全关闭")
# 使用系统播放器
self.play_with_system_player(filename)
print("🔄 准备重新开启音频输入")
except Exception as e:
print(f"❌ 播放失败: {e}")
finally:
# 恢复播放状态
self.is_playing = False
# 等待播放完全结束
time.sleep(0.5)
# 重新开启输入流
self._setup_audio()
# 重置所有状态
self.energy_history = []
self.zcr_history = []
print("📡 音频输入已重新开启")
def update_pre_record_buffer(self, audio_data):
"""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"""
print("🎙️ 检测到声音，开始录音...")
self.recording = True
self.recorded_frames = []
# 将预录音缓冲区的内容添加到录音中
self.recorded_frames.extend(self.pre_record_buffer)
# 清空预录音缓冲区
self.pre_record_buffer = []
self.recording_start_time = time.time()
self.last_sound_time = time.time()
self.energy_history = []
self.zcr_history = [] # 重置ZCR历史
# 重置ZCR相关计数器
self.consecutive_low_zcr_count = 0
self.consecutive_silence_count = 0
self.voice_activity_history = []
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) # 16位音频，每样本2字节
# 计算实际录音时长和预录音时长
actual_duration = duration
pre_record_duration = min(duration, self.pre_record_duration)
print(f"📝 录音完成，时长: {actual_duration:.2f}秒 (包含预录音 {pre_record_duration:.1f}秒)")
# 保存录音
success, filename = self.save_recording(audio_data)
# 如果保存成功，播放录音
if success and filename:
print("=" * 50)
print("🔊 播放刚才录制的音频...")
# 优先使用系统播放器避免回声
self.play_audio_safe(filename)
print("=" * 50)
self.recording = False
self.recorded_frames = []
self.recording_start_time = None
self.last_sound_time = None
self.energy_history = []
self.zcr_history = []
def monitor_performance(self):
"""性能监控"""
self.frame_count += 1
if self.frame_count % 1000 == 0: # 每1000帧显示一次
elapsed = time.time() - self.start_time
fps = self.frame_count / elapsed
avg_energy = self.get_average_energy()
print(f"📊 性能: {fps:.1f} FPS | 平均能量: {avg_energy:.1f} | 阈值: {self.energy_threshold}")
def auto_adjust_threshold(self):
"""自动调整能量阈值"""
if len(self.energy_history) >= 20:
# 基于历史能量的中位数和标准差调整阈值
median_energy = np.median(self.energy_history)
std_energy = np.std(self.energy_history)
# 设置阈值为中位数 + 2倍标准差
new_threshold = max(300, median_energy + 2 * std_energy)
# 平滑调整阈值
self.energy_threshold = 0.9 * self.energy_threshold + 0.1 * new_threshold
def run(self):
"""运行录音系统"""
if not self.stream:
print("❌ 音频设备未初始化")
return
self.running = True
print("🎤 开始监听...")
print(f"能量阈值: {self.energy_threshold} (已弃用)")
print(f"静音阈值: {self.silence_threshold}秒")
print("📖 使用说明:")
print("- 检测到声音自动开始录音")
print("- 持续静音3秒自动结束录音")
print("- 最少录音2秒，最多30秒")
print("- 录音完成后自动播放")
print("- 按 Ctrl+C 退出")
print("🎯 新增功能:")
print("- 纯ZCR语音检测（移除能量检测）")
print("- 零交叉率检测（区分语音和噪音）")
print("- 实时显示ZCR状态")
print("- 预录音功能（包含声音开始前2秒）")
print("- 环形缓冲区防止丢失开头音频")
print("🤖 纯ZCR静音检测:")
print("- 连续低ZCR计数（20次=2秒）")
print("- ZCR活动历史追踪")
print("- 基于ZCR模式的静音验证")
print("- 语音范围: 1200-6000 Hz (提高阈值)")
print("=" * 50)
try:
while self.running:
# 读取音频数据
data = self.stream.read(self.CHUNK_SIZE, exception_on_overflow=False)
if len(data) == 0:
continue
# 如果正在播放，完全跳过音频处理
if self.is_playing:
# 显示播放状态
status = "🔊 播放中... 跳过录音处理"
print(f"\r{status}", end='', flush=True)
time.sleep(0.1) # 播放时增加延迟减少CPU使用
continue
# 计算能量和零交叉率
energy = self.calculate_energy(data)
zcr = self.calculate_zero_crossing_rate(data)
peak_energy = self.calculate_peak_energy(data)
# 性能监控
self.monitor_performance()
if self.recording:
# 录音模式
self.recorded_frames.append(data)
recording_duration = time.time() - self.recording_start_time
# 基于ZCR的智能静音检测
if self.is_voice_active_advanced(energy, zcr):
self.last_sound_time = time.time()
self.consecutive_low_zcr_count = 0 # 重置低ZCR计数
self.consecutive_silence_count = 0 # 重置静音计数
else:
self.consecutive_low_zcr_count += 1 # 增加低ZCR计数
self.consecutive_silence_count += 1 # 增加静音计数
# 更新ZCR活动历史（基于ZCR是否在语音范围内）
self.voice_activity_history.append(1200 < zcr < 6000)
if len(self.voice_activity_history) > self.max_voice_history:
self.voice_activity_history.pop(0)
# 检查是否应该结束录音
current_time = time.time()
# 纯ZCR静音检测
should_stop = False
stop_reason = ""
# 主要检测：连续低ZCR计数
if self.consecutive_low_zcr_count >= self.low_zcr_threshold_count:
# 进一步验证：检查最近的ZCR活动历史
if len(self.voice_activity_history) >= 15:
recent_voice_activity = sum(self.voice_activity_history[-15:])
if recent_voice_activity <= 3: # 最近15个样本中最多3个有语音活动
should_stop = True
stop_reason = f"ZCR静音检测 ({self.consecutive_low_zcr_count}次连续低ZCR)"
else:
# 如果历史数据不足，使用基础检测
should_stop = True
stop_reason = f"基础ZCR静音检测 ({self.consecutive_low_zcr_count}次)"
# 备用检测：基于时间的静音检测
if not should_stop and current_time - self.last_sound_time > self.silence_threshold:
should_stop = True
stop_reason = f"时间静音检测 ({self.silence_threshold}秒)"
# 执行停止录音
if should_stop and recording_duration >= self.min_recording_time:
print(f"\n🔇 {stop_reason}，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.max_recording_time}秒")
self.stop_recording()
# 显示录音状态（仅ZCR相关信息）
is_voice = self.is_voice_active_advanced(energy, zcr)
zcr_progress = f"{self.consecutive_low_zcr_count}/{self.low_zcr_threshold_count}"
recent_activity = sum(self.voice_activity_history[-5:]) if len(self.voice_activity_history) >= 5 else 0
status = f"录音中... {recording_duration:.1f}s | ZCR: {zcr:.0f} | 语音: {is_voice} | 低ZCR计数: {zcr_progress} | 活动: {recent_activity}"
print(f"\r{status}", end='', flush=True)
else:
# 监听模式 - 更新预录音缓冲区
self.update_pre_record_buffer(data)
# 使用高级检测
if self.is_voice_active_advanced(energy, zcr):
# 检测到声音，开始录音
self.start_recording()
else:
# 显示监听状态（仅ZCR相关信息）
is_voice = self.is_voice_active_advanced(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)
# 减少CPU使用
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
if self.recording:
self.stop_recording()
if self.stream:
self.stream.stop_stream()
self.stream.close()
if self.audio:
self.audio.terminate()
def main():
"""主函数"""
print("🚀 基于能量检测的极简录音系统")
print("=" * 50)
# 创建录音系统
recorder = EnergyBasedRecorder(
energy_threshold=200, # 能量阈值（降低以提高灵敏度）
silence_threshold=3.0, # 静音阈值（秒）- 改为3秒
min_recording_time=2.0, # 最小录音时间
max_recording_time=30.0 # 最大录音时间
)
print("✅ 系统初始化成功")
print("🎯 优化特点:")
print(" - 完全移除Vosk识别依赖")
print(" - 基于能量检测，极低CPU占用")
print(" - 自动调整能量阈值")
print(" - 实时性能监控")
print(" - 预期延迟: <0.1秒")
print("=" * 50)
# 开始运行
recorder.run()
if __name__ == "__main__":
main()

4
requirements.txt
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@ -1,3 +1,5 @@
vosk>=0.3.44
pyaudio>=0.2.11
numpy>=1.19.0
numpy>=1.19.0
aiohttp>=3.8.0
asyncio

15
sauc_python/readme.md Normal file
View File

@ -0,0 +1,15 @@
# README
**asr tob 相关client demo**
# Notice
python version: python 3.x
替换代码中的key为真实数据:
"app_key": "xxxxxxx",
"access_key": "xxxxxxxxxxxxxxxx"
使用示例：
python3 sauc_websocket_demo.py --file /Users/bytedance/code/python/eng_ddc_itn.wav

523
sauc_python/sauc_websocket_demo.py Normal file
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@ -0,0 +1,523 @@
import asyncio
import aiohttp
import json
import struct
import gzip
import uuid
import logging
import os
import subprocess
from typing import Optional, List, Dict, Any, Tuple, AsyncGenerator
# 配置日志
logging.basicConfig(
level=logging.INFO,
format='%(asctime)s - %(levelname)s - %(message)s',
handlers=[
logging.FileHandler('run.log'),
logging.StreamHandler()
]
)
logger = logging.getLogger(__name__)
# 常量定义
DEFAULT_SAMPLE_RATE = 16000
class ProtocolVersion:
V1 = 0b0001
class MessageType:
CLIENT_FULL_REQUEST = 0b0001
CLIENT_AUDIO_ONLY_REQUEST = 0b0010
SERVER_FULL_RESPONSE = 0b1001
SERVER_ERROR_RESPONSE = 0b1111
class MessageTypeSpecificFlags:
NO_SEQUENCE = 0b0000
POS_SEQUENCE = 0b0001
NEG_SEQUENCE = 0b0010
NEG_WITH_SEQUENCE = 0b0011
class SerializationType:
NO_SERIALIZATION = 0b0000
JSON = 0b0001
class CompressionType:
GZIP = 0b0001
class Config:
def __init__(self):
# 填入控制台获取的app id和access token
self.auth = {
"app_key": "xxxxxxx",
"access_key": "xxxxxxxxxxxx"
}
@property
def app_key(self) -> str:
return self.auth["app_key"]
@property
def access_key(self) -> str:
return self.auth["access_key"]
config = Config()
class CommonUtils:
@staticmethod
def gzip_compress(data: bytes) -> bytes:
return gzip.compress(data)
@staticmethod
def gzip_decompress(data: bytes) -> bytes:
return gzip.decompress(data)
@staticmethod
def judge_wav(data: bytes) -> bool:
if len(data) < 44:
return False
return data[:4] == b'RIFF' and data[8:12] == b'WAVE'
@staticmethod
def convert_wav_with_path(audio_path: str, sample_rate: int = DEFAULT_SAMPLE_RATE) -> bytes:
try:
cmd = [
"ffmpeg", "-v", "quiet", "-y", "-i", audio_path,
"-acodec", "pcm_s16le", "-ac", "1", "-ar", str(sample_rate),
"-f", "wav", "-"
]
result = subprocess.run(cmd, check=True, stdout=subprocess.PIPE, stderr=subprocess.PIPE)
# 尝试删除原始文件
try:
os.remove(audio_path)
except OSError as e:
logger.warning(f"Failed to remove original file: {e}")
return result.stdout
except subprocess.CalledProcessError as e:
logger.error(f"FFmpeg conversion failed: {e.stderr.decode()}")
raise RuntimeError(f"Audio conversion failed: {e.stderr.decode()}")
@staticmethod
def read_wav_info(data: bytes) -> Tuple[int, int, int, int, bytes]:
if len(data) < 44:
raise ValueError("Invalid WAV file: too short")
# 解析WAV头
chunk_id = data[:4]
if chunk_id != b'RIFF':
raise ValueError("Invalid WAV file: not RIFF format")
format_ = data[8:12]
if format_ != b'WAVE':
raise ValueError("Invalid WAV file: not WAVE format")
# 解析fmt子块
audio_format = struct.unpack('<H', data[20:22])[0]
num_channels = struct.unpack('<H', data[22:24])[0]
sample_rate = struct.unpack('<I', data[24:28])[0]
bits_per_sample = struct.unpack('<H', data[34:36])[0]
# 查找data子块
pos = 36
while pos < len(data) - 8:
subchunk_id = data[pos:pos+4]
subchunk_size = struct.unpack('<I', data[pos+4:pos+8])[0]
if subchunk_id == b'data':
wave_data = data[pos+8:pos+8+subchunk_size]
return (
num_channels,
bits_per_sample // 8,
sample_rate,
subchunk_size // (num_channels * (bits_per_sample // 8)),
wave_data
)
pos += 8 + subchunk_size
raise ValueError("Invalid WAV file: no data subchunk found")
class AsrRequestHeader:
def __init__(self):
self.message_type = MessageType.CLIENT_FULL_REQUEST
self.message_type_specific_flags = MessageTypeSpecificFlags.POS_SEQUENCE
self.serialization_type = SerializationType.JSON
self.compression_type = CompressionType.GZIP
self.reserved_data = bytes([0x00])
def with_message_type(self, message_type: int) -> 'AsrRequestHeader':
self.message_type = message_type
return self
def with_message_type_specific_flags(self, flags: int) -> 'AsrRequestHeader':
self.message_type_specific_flags = flags
return self
def with_serialization_type(self, serialization_type: int) -> 'AsrRequestHeader':
self.serialization_type = serialization_type
return self
def with_compression_type(self, compression_type: int) -> 'AsrRequestHeader':
self.compression_type = compression_type
return self
def with_reserved_data(self, reserved_data: bytes) -> 'AsrRequestHeader':
self.reserved_data = reserved_data
return self
def to_bytes(self) -> bytes:
header = bytearray()
header.append((ProtocolVersion.V1 << 4) | 1)
header.append((self.message_type << 4) | self.message_type_specific_flags)
header.append((self.serialization_type << 4) | self.compression_type)
header.extend(self.reserved_data)
return bytes(header)
@staticmethod
def default_header() -> 'AsrRequestHeader':
return AsrRequestHeader()
class RequestBuilder:
@staticmethod
def new_auth_headers() -> Dict[str, str]:
reqid = str(uuid.uuid4())
return {
"X-Api-Resource-Id": "volc.bigasr.sauc.duration",
"X-Api-Request-Id": reqid,
"X-Api-Access-Key": config.access_key,
"X-Api-App-Key": config.app_key
}
@staticmethod
def new_full_client_request(seq: int) -> bytes: # 添加seq参数
header = AsrRequestHeader.default_header() \
.with_message_type_specific_flags(MessageTypeSpecificFlags.POS_SEQUENCE)
payload = {
"user": {
"uid": "demo_uid"
},
"audio": {
"format": "wav",
"codec": "raw",
"rate": 16000,
"bits": 16,
"channel": 1
},
"request": {
"model_name": "bigmodel",
"enable_itn": True,
"enable_punc": True,
"enable_ddc": True,
"show_utterances": True,
"enable_nonstream": False
}
}
payload_bytes = json.dumps(payload).encode('utf-8')
compressed_payload = CommonUtils.gzip_compress(payload_bytes)
payload_size = len(compressed_payload)
request = bytearray()
request.extend(header.to_bytes())
request.extend(struct.pack('>i', seq)) # 使用传入的seq
request.extend(struct.pack('>I', payload_size))
request.extend(compressed_payload)
return bytes(request)
@staticmethod
def new_audio_only_request(seq: int, segment: bytes, is_last: bool = False) -> bytes:
header = AsrRequestHeader.default_header()
if is_last: # 最后一个包特殊处理
header.with_message_type_specific_flags(MessageTypeSpecificFlags.NEG_WITH_SEQUENCE)
seq = -seq # 设为负值
else:
header.with_message_type_specific_flags(MessageTypeSpecificFlags.POS_SEQUENCE)
header.with_message_type(MessageType.CLIENT_AUDIO_ONLY_REQUEST)
request = bytearray()
request.extend(header.to_bytes())
request.extend(struct.pack('>i', seq))
compressed_segment = CommonUtils.gzip_compress(segment)
request.extend(struct.pack('>I', len(compressed_segment)))
request.extend(compressed_segment)
return bytes(request)
class AsrResponse:
def __init__(self):
self.code = 0
self.event = 0
self.is_last_package = False
self.payload_sequence = 0
self.payload_size = 0
self.payload_msg = None
def to_dict(self) -> Dict[str, Any]:
return {
"code": self.code,
"event": self.event,
"is_last_package": self.is_last_package,
"payload_sequence": self.payload_sequence,
"payload_size": self.payload_size,
"payload_msg": self.payload_msg
}
class ResponseParser:
@staticmethod
def parse_response(msg: bytes) -> AsrResponse:
response = AsrResponse()
header_size = msg[0] & 0x0f
message_type = msg[1] >> 4
message_type_specific_flags = msg[1] & 0x0f
serialization_method = msg[2] >> 4
message_compression = msg[2] & 0x0f
payload = msg[header_size*4:]
# 解析message_type_specific_flags
if message_type_specific_flags & 0x01:
response.payload_sequence = struct.unpack('>i', payload[:4])[0]
payload = payload[4:]
if message_type_specific_flags & 0x02:
response.is_last_package = True
if message_type_specific_flags & 0x04:
response.event = struct.unpack('>i', payload[:4])[0]
payload = payload[4:]
# 解析message_type
if message_type == MessageType.SERVER_FULL_RESPONSE:
response.payload_size = struct.unpack('>I', payload[:4])[0]
payload = payload[4:]
elif message_type == MessageType.SERVER_ERROR_RESPONSE:
response.code = struct.unpack('>i', payload[:4])[0]
response.payload_size = struct.unpack('>I', payload[4:8])[0]
payload = payload[8:]
if not payload:
return response
# 解压缩
if message_compression == CompressionType.GZIP:
try:
payload = CommonUtils.gzip_decompress(payload)
except Exception as e:
logger.error(f"Failed to decompress payload: {e}")
return response
# 解析payload
try:
if serialization_method == SerializationType.JSON:
response.payload_msg = json.loads(payload.decode('utf-8'))
except Exception as e:
logger.error(f"Failed to parse payload: {e}")
return response
class AsrWsClient:
def __init__(self, url: str, segment_duration: int = 200):
self.seq = 1
self.url = url
self.segment_duration = segment_duration
self.conn = None
self.session = None # 添加session引用
async def __aenter__(self):
self.session = aiohttp.ClientSession()
return self
async def __aexit__(self, exc_type, exc, tb):
if self.conn and not self.conn.closed:
await self.conn.close()
if self.session and not self.session.closed:
await self.session.close()
async def read_audio_data(self, file_path: str) -> bytes:
try:
with open(file_path, 'rb') as f:
content = f.read()
if not CommonUtils.judge_wav(content):
logger.info("Converting audio to WAV format...")
content = CommonUtils.convert_wav_with_path(file_path, DEFAULT_SAMPLE_RATE)
return content
except Exception as e:
logger.error(f"Failed to read audio data: {e}")
raise
def get_segment_size(self, content: bytes) -> int:
try:
channel_num, samp_width, frame_rate, _, _ = CommonUtils.read_wav_info(content)[:5]
size_per_sec = channel_num * samp_width * frame_rate
segment_size = size_per_sec * self.segment_duration // 1000
return segment_size
except Exception as e:
logger.error(f"Failed to calculate segment size: {e}")
raise
async def create_connection(self) -> None:
headers = RequestBuilder.new_auth_headers()
try:
self.conn = await self.session.ws_connect( # 使用self.session
self.url,
headers=headers
)
logger.info(f"Connected to {self.url}")
except Exception as e:
logger.error(f"Failed to connect to WebSocket: {e}")
raise
async def send_full_client_request(self) -> None:
request = RequestBuilder.new_full_client_request(self.seq)
self.seq += 1 # 发送后递增
try:
await self.conn.send_bytes(request)
logger.info(f"Sent full client request with seq: {self.seq-1}")
msg = await self.conn.receive()
if msg.type == aiohttp.WSMsgType.BINARY:
response = ResponseParser.parse_response(msg.data)
logger.info(f"Received response: {response.to_dict()}")
else:
logger.error(f"Unexpected message type: {msg.type}")
except Exception as e:
logger.error(f"Failed to send full client request: {e}")
raise
async def send_messages(self, segment_size: int, content: bytes) -> AsyncGenerator[None, None]:
audio_segments = self.split_audio(content, segment_size)
total_segments = len(audio_segments)
for i, segment in enumerate(audio_segments):
is_last = (i == total_segments - 1)
request = RequestBuilder.new_audio_only_request(
self.seq,
segment,
is_last=is_last
)
await self.conn.send_bytes(request)
logger.info(f"Sent audio segment with seq: {self.seq} (last: {is_last})")
if not is_last:
self.seq += 1
await asyncio.sleep(self.segment_duration / 1000) # 逐个发送，间隔时间模拟实时流
# 让出控制权，允许接受消息
yield
async def recv_messages(self) -> AsyncGenerator[AsrResponse, None]:
try:
async for msg in self.conn:
if msg.type == aiohttp.WSMsgType.BINARY:
response = ResponseParser.parse_response(msg.data)
yield response
if response.is_last_package or response.code != 0:
break
elif msg.type == aiohttp.WSMsgType.ERROR:
logger.error(f"WebSocket error: {msg.data}")
break
elif msg.type == aiohttp.WSMsgType.CLOSED:
logger.info("WebSocket connection closed")
break
except Exception as e:
logger.error(f"Error receiving messages: {e}")
raise
async def start_audio_stream(self, segment_size: int, content: bytes) -> AsyncGenerator[AsrResponse, None]:
async def sender():
async for _ in self.send_messages(segment_size, content):
pass
# 启动发送和接收任务
sender_task = asyncio.create_task(sender())
try:
async for response in self.recv_messages():
yield response
finally:
sender_task.cancel()
try:
await sender_task
except asyncio.CancelledError:
pass
@staticmethod
def split_audio(data: bytes, segment_size: int) -> List[bytes]:
if segment_size <= 0:
return []
segments = []
for i in range(0, len(data), segment_size):
end = i + segment_size
if end > len(data):
end = len(data)
segments.append(data[i:end])
return segments
async def execute(self, file_path: str) -> AsyncGenerator[AsrResponse, None]:
if not file_path:
raise ValueError("File path is empty")
if not self.url:
raise ValueError("URL is empty")
self.seq = 1
try:
# 1. 读取音频文件
content = await self.read_audio_data(file_path)
# 2. 计算分段大小
segment_size = self.get_segment_size(content)
# 3. 创建WebSocket连接
await self.create_connection()
# 4. 发送完整客户端请求
await self.send_full_client_request()
# 5. 启动音频流处理
async for response in self.start_audio_stream(segment_size, content):
yield response
except Exception as e:
logger.error(f"Error in ASR execution: {e}")
raise
finally:
if self.conn:
await self.conn.close()
async def main():
import argparse
parser = argparse.ArgumentParser(description="ASR WebSocket Client")
parser.add_argument("--file", type=str, required=True, help="Audio file path")
#wss://openspeech.bytedance.com/api/v3/sauc/bigmodel
#wss://openspeech.bytedance.com/api/v3/sauc/bigmodel_async
#wss://openspeech.bytedance.com/api/v3/sauc/bigmodel_nostream
parser.add_argument("--url", type=str, default="wss://openspeech.bytedance.com/api/v3/sauc/bigmodel_nostream",
help="WebSocket URL")
parser.add_argument("--seg-duration", type=int, default=200,
help="Audio duration(ms) per packet, default:200")
args = parser.parse_args()
async with AsrWsClient(args.url, args.seg_duration) as client: # 使用async with
try:
async for response in client.execute(args.file):
logger.info(f"Received response: {json.dumps(response.to_dict(), indent=2, ensure_ascii=False)}")
except Exception as e:
logger.error(f"ASR processing failed: {e}")
if __name__ == "__main__":
asyncio.run(main())
# 用法：
# python3 sauc_websocket_demo.py --file /Users/bytedance/code/python/eng_ddc_itn.wav

403
simple_wake_and_record.py
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@ -1,403 +0,0 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
简化的唤醒+录音测试
专注于解决音频冲突问题
"""
import sys
import os
import time
import threading
import pyaudio
import json
# 添加当前目录到路径
sys.path.append(os.path.dirname(os.path.abspath(__file__)))
try:
from vosk import Model, KaldiRecognizer
VOSK_AVAILABLE = True
except ImportError:
VOSK_AVAILABLE = False
print("⚠️ Vosk 未安装，请运行: pip install vosk")
class SimpleWakeAndRecord:
"""简化的唤醒+录音系统"""
def __init__(self, model_path="model", wake_words=["你好", "助手"]):
self.model_path = model_path
self.wake_words = wake_words
self.model = None
self.recognizer = None
self.audio = None
self.stream = None
self.running = False
# 音频参数
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
self.RATE = 16000
self.CHUNK_SIZE = 1024
# 录音相关
self.recording = False
self.recorded_frames = []
self.last_text_time = None # 最后一次识别到文字的时间
self.recording_start_time = None
self.recording_recognizer = None # 录音时专用的识别器
# 阈值
self.text_silence_threshold = 3.0 # 3秒没有识别到文字就结束
self.min_recording_time = 2.0 # 最小录音时间
self.max_recording_time = 30.0 # 最大录音时间
self._setup_model()
self._setup_audio()
def _setup_model(self):
"""设置 Vosk 模型"""
if not VOSK_AVAILABLE:
return
try:
if not os.path.exists(self.model_path):
print(f"模型路径不存在: {self.model_path}")
return
self.model = Model(self.model_path)
self.recognizer = KaldiRecognizer(self.model, self.RATE)
self.recognizer.SetWords(True)
print(f"✅ Vosk 模型加载成功")
except Exception as e:
print(f"模型初始化失败: {e}")
def _setup_audio(self):
"""设置音频设备"""
try:
if self.audio is None:
self.audio = pyaudio.PyAudio()
if self.stream is None:
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 _calculate_energy(self, audio_data):
"""计算音频能量"""
if len(audio_data) == 0:
return 0
import numpy as np
audio_array = np.frombuffer(audio_data, dtype=np.int16)
rms = np.sqrt(np.mean(audio_array ** 2))
return rms
def _check_wake_word(self, text):
"""检查是否包含唤醒词"""
if not text or not self.wake_words:
return False, None
text_lower = text.lower()
for wake_word in self.wake_words:
if wake_word.lower() in text_lower:
return True, wake_word
return False, None
def _save_recording(self, audio_data):
"""保存录音"""
timestamp = time.strftime("%Y%m%d_%H%M%S")
filename = f"recording_{timestamp}.wav"
try:
import wave
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:
import wave
# 打开音频文件
with wave.open(filename, 'rb') as wf:
# 获取音频参数
channels = wf.getnchannels()
width = wf.getsampwidth()
rate = wf.getframerate()
total_frames = wf.getnframes()
# 分块读取音频数据，避免内存问题
chunk_size = 1024
frames = []
for _ in range(0, total_frames, chunk_size):
chunk = wf.readframes(chunk_size)
if chunk:
frames.append(chunk)
else:
break
# 创建播放流
playback_stream = self.audio.open(
format=self.audio.get_format_from_width(width),
channels=channels,
rate=rate,
output=True
)
print(f"🔊 开始播放: {filename}")
# 分块播放音频
for chunk in frames:
playback_stream.write(chunk)
# 等待播放完成
playback_stream.stop_stream()
playback_stream.close()
print("✅ 播放完成")
except Exception as e:
print(f"❌ 播放失败: {e}")
# 如果pyaudio播放失败，尝试用系统命令播放
self._play_with_system_player(filename)
def _play_with_system_player(self, filename):
"""使用系统播放器播放音频"""
try:
import platform
import subprocess
system = platform.system()
if system == 'Darwin': # macOS
cmd = ['afplay', filename]
elif system == 'Windows':
cmd = ['start', '/min', filename]
else: # Linux
cmd = ['aplay', filename]
print(f"🔊 使用系统播放器: {' '.join(cmd)}")
subprocess.run(cmd, check=True)
print("✅ 播放完成")
except Exception as e:
print(f"❌ 系统播放器也失败: {e}")
print(f"💡 文件已保存，请手动播放: {filename}")
def _start_recording(self):
"""开始录音"""
print("🎙️ 开始录音，请说话...")
self.recording = True
self.recorded_frames = []
self.last_text_time = None
self.recording_start_time = time.time()
# 为录音创建一个新的识别器
if self.model:
self.recording_recognizer = KaldiRecognizer(self.model, self.RATE)
self.recording_recognizer.SetWords(True)
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}秒")
# 保存录音
success, filename = self._save_recording(audio_data)
# 如果保存成功，播放录音
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.last_text_time = None
self.recording_start_time = None
self.recording_recognizer = None
def start(self):
"""开始唤醒词检测和录音"""
if not self.stream:
print("❌ 音频设备未初始化")
return
self.running = True
print("🎤 开始监听...")
print(f"唤醒词: {', '.join(self.wake_words)}")
try:
while self.running:
# 读取音频数据
data = self.stream.read(self.CHUNK_SIZE, exception_on_overflow=False)
if len(data) == 0:
continue
if self.recording:
# 录音模式
self.recorded_frames.append(data)
recording_duration = time.time() - self.recording_start_time
# 使用录音专用的识别器进行实时识别
if self.recording_recognizer:
if self.recording_recognizer.AcceptWaveform(data):
# 获取最终识别结果
result = json.loads(self.recording_recognizer.Result())
text = result.get('text', '').strip()
if text:
# 识别到文字，更新时间戳
self.last_text_time = time.time()
print(f"\n📝 识别: {text}")
else:
# 获取部分识别结果
partial_result = json.loads(self.recording_recognizer.PartialResult())
partial_text = partial_result.get('partial', '').strip()
if partial_text:
# 更新时间戳（部分识别也算有声音）
self.last_text_time = time.time()
status = f"录音中... {recording_duration:.1f}s | {partial_text}"
print(f"\r{status}", end='', flush=True)
# 检查是否需要结束录音
current_time = time.time()
# 检查是否有文字识别超时
if self.last_text_time is not None:
text_silence_duration = current_time - self.last_text_time
if text_silence_duration > self.text_silence_threshold and recording_duration >= self.min_recording_time:
print(f"\n\n3秒没有识别到文字，结束录音")
self._stop_recording()
else:
# 还没有识别到任何文字，检查是否超时
if recording_duration > 5.0: # 如果5秒还没识别到任何文字，也结束
print(f"\n\n5秒没有识别到文字，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\n达到最大录音时间 {self.max_recording_time}s")
self._stop_recording()
# 显示录音状态
if self.last_text_time is None:
status = f"等待语音输入... {recording_duration:.1f}s"
print(f"\r{status}", end='', flush=True)
elif self.model and self.recognizer:
# 唤醒词检测模式
if self.recognizer.AcceptWaveform(data):
result = json.loads(self.recognizer.Result())
text = result.get('text', '').strip()
if text:
print(f"识别: {text}")
# 检查唤醒词
is_wake_word, detected_word = self._check_wake_word(text)
if is_wake_word:
print(f"🎯 检测到唤醒词: {detected_word}")
self._start_recording()
else:
# 显示实时音频级别
energy = self._calculate_energy(data)
if energy > 50: # 只显示有意义的音频级别
partial_result = json.loads(self.recognizer.PartialResult())
partial_text = partial_result.get('partial', '')
if partial_text:
status = f"监听中... 能量: {energy:.0f} | {partial_text}"
else:
status = f"监听中... 能量: {energy:.0f}"
print(status, end='\r')
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"""
self.running = False
if self.recording:
self._stop_recording()
if self.stream:
self.stream.stop_stream()
self.stream.close()
self.stream = None
if self.audio:
self.audio.terminate()
self.audio = None
def main():
"""主函数"""
print("🚀 简化唤醒+录音测试")
print("=" * 50)
# 检查模型
model_dir = "model"
if not os.path.exists(model_dir):
print("⚠️ 未找到模型目录")
print("请下载 Vosk 模型到 model 目录")
return
# 创建系统
system = SimpleWakeAndRecord(
model_path=model_dir,
wake_words=["你好", "助手", "小爱"]
)
if not system.model:
print("❌ 模型加载失败")
return
print("✅ 系统初始化成功")
print("📖 使用说明:")
print("1. 说唤醒词开始录音")
print("2. 基于语音识别判断，3秒没有识别到文字就结束")
print("3. 最少录音2秒，最多30秒")
print("4. 录音时实时显示识别结果")
print("5. 录音文件自动保存")
print("6. 录音完成后自动播放刚才录制的内容")
print("7. 按 Ctrl+C 退出")
print("=" * 50)
# 开始运行
system.start()
if __name__ == "__main__":
main()

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speech_recognizer.py Normal file
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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
语音识别模块
基于 SAUC API 为录音文件提供语音识别功能
"""
import os
import json
import time
import logging
import asyncio
import aiohttp
import struct
import gzip
import uuid
from typing import Optional, List, Dict, Any, AsyncGenerator
from dataclasses import dataclass
# 配置日志
logging.basicConfig(
level=logging.INFO,
format='%(asctime)s - %(levelname)s - %(message)s'
)
logger = logging.getLogger(__name__)
# 常量定义
DEFAULT_SAMPLE_RATE = 16000
class ProtocolVersion:
V1 = 0b0001
class MessageType:
CLIENT_FULL_REQUEST = 0b0001
CLIENT_AUDIO_ONLY_REQUEST = 0b0010
SERVER_FULL_RESPONSE = 0b1001
SERVER_ERROR_RESPONSE = 0b1111
class MessageTypeSpecificFlags:
NO_SEQUENCE = 0b0000
POS_SEQUENCE = 0b0001
NEG_SEQUENCE = 0b0010
NEG_WITH_SEQUENCE = 0b0011
class SerializationType:
NO_SERIALIZATION = 0b0000
JSON = 0b0001
class CompressionType:
GZIP = 0b0001
@dataclass
class RecognitionResult:
"""语音识别结果"""
text: str
confidence: float
is_final: bool
start_time: Optional[float] = None
end_time: Optional[float] = None
class AudioUtils:
"""音频处理工具类"""
@staticmethod
def gzip_compress(data: bytes) -> bytes:
"""GZIP压缩"""
return gzip.compress(data)
@staticmethod
def gzip_decompress(data: bytes) -> bytes:
"""GZIP解压缩"""
return gzip.decompress(data)
@staticmethod
def is_wav_file(data: bytes) -> bool:
"""检查是否为WAV文件"""
if len(data) < 44:
return False
return data[:4] == b'RIFF' and data[8:12] == b'WAVE'
@staticmethod
def read_wav_info(data: bytes) -> tuple:
"""读取WAV文件信息"""
if len(data) < 44:
raise ValueError("Invalid WAV file: too short")
# 解析WAV头
chunk_id = data[:4]
if chunk_id != b'RIFF':
raise ValueError("Invalid WAV file: not RIFF format")
format_ = data[8:12]
if format_ != b'WAVE':
raise ValueError("Invalid WAV file: not WAVE format")
# 解析fmt子块
audio_format = struct.unpack('<H', data[20:22])[0]
num_channels = struct.unpack('<H', data[22:24])[0]
sample_rate = struct.unpack('<I', data[24:28])[0]
bits_per_sample = struct.unpack('<H', data[34:36])[0]
# 查找data子块
pos = 36
while pos < len(data) - 8:
subchunk_id = data[pos:pos+4]
subchunk_size = struct.unpack('<I', data[pos+4:pos+8])[0]
if subchunk_id == b'data':
wave_data = data[pos+8:pos+8+subchunk_size]
return (
num_channels,
bits_per_sample // 8,
sample_rate,
subchunk_size // (num_channels * (bits_per_sample // 8)),
wave_data
)
pos += 8 + subchunk_size
raise ValueError("Invalid WAV file: no data subchunk found")
class AsrConfig:
"""ASR配置"""
def __init__(self, app_key: str = None, access_key: str = None):
self.auth = {
"app_key": app_key or os.getenv("SAUC_APP_KEY", "your_app_key"),
"access_key": access_key or os.getenv("SAUC_ACCESS_KEY", "your_access_key")
}
@property
def app_key(self) -> str:
return self.auth["app_key"]
@property
def access_key(self) -> str:
return self.auth["access_key"]
class AsrRequestHeader:
"""ASR请求头"""
def __init__(self):
self.message_type = MessageType.CLIENT_FULL_REQUEST
self.message_type_specific_flags = MessageTypeSpecificFlags.POS_SEQUENCE
self.serialization_type = SerializationType.JSON
self.compression_type = CompressionType.GZIP
self.reserved_data = bytes([0x00])
def with_message_type(self, message_type: int) -> 'AsrRequestHeader':
self.message_type = message_type
return self
def with_message_type_specific_flags(self, flags: int) -> 'AsrRequestHeader':
self.message_type_specific_flags = flags
return self
def with_serialization_type(self, serialization_type: int) -> 'AsrRequestHeader':
self.serialization_type = serialization_type
return self
def with_compression_type(self, compression_type: int) -> 'AsrRequestHeader':
self.compression_type = compression_type
return self
def with_reserved_data(self, reserved_data: bytes) -> 'AsrRequestHeader':
self.reserved_data = reserved_data
return self
def to_bytes(self) -> bytes:
header = bytearray()
header.append((ProtocolVersion.V1 << 4) | 1)
header.append((self.message_type << 4) | self.message_type_specific_flags)
header.append((self.serialization_type << 4) | self.compression_type)
header.extend(self.reserved_data)
return bytes(header)
@staticmethod
def default_header() -> 'AsrRequestHeader':
return AsrRequestHeader()
class RequestBuilder:
"""请求构建器"""
@staticmethod
def new_auth_headers(config: AsrConfig) -> Dict[str, str]:
"""创建认证头"""
reqid = str(uuid.uuid4())
return {
"X-Api-Resource-Id": "volc.bigasr.sauc.duration",
"X-Api-Request-Id": reqid,
"X-Api-Access-Key": config.access_key,
"X-Api-App-Key": config.app_key
}
@staticmethod
def new_full_client_request(seq: int) -> bytes:
"""创建完整客户端请求"""
header = AsrRequestHeader.default_header() \
.with_message_type_specific_flags(MessageTypeSpecificFlags.POS_SEQUENCE)
payload = {
"user": {
"uid": "local_voice_user"
},
"audio": {
"format": "wav",
"codec": "raw",
"rate": 16000,
"bits": 16,
"channel": 1
},
"request": {
"model_name": "bigmodel",
"enable_itn": True,
"enable_punc": True,
"enable_ddc": True,
"show_utterances": True,
"enable_nonstream": False
}
}
payload_bytes = json.dumps(payload).encode('utf-8')
compressed_payload = AudioUtils.gzip_compress(payload_bytes)
payload_size = len(compressed_payload)
request = bytearray()
request.extend(header.to_bytes())
request.extend(struct.pack('>i', seq))
request.extend(struct.pack('>U', payload_size))
request.extend(compressed_payload)
return bytes(request)
@staticmethod
def new_audio_only_request(seq: int, segment: bytes, is_last: bool = False) -> bytes:
"""创建纯音频请求"""
header = AsrRequestHeader.default_header()
if is_last:
header.with_message_type_specific_flags(MessageTypeSpecificFlags.NEG_WITH_SEQUENCE)
seq = -seq
else:
header.with_message_type_specific_flags(MessageTypeSpecificFlags.POS_SEQUENCE)
header.with_message_type(MessageType.CLIENT_AUDIO_ONLY_REQUEST)
request = bytearray()
request.extend(header.to_bytes())
request.extend(struct.pack('>i', seq))
compressed_segment = AudioUtils.gzip_compress(segment)
request.extend(struct.pack('>U', len(compressed_segment)))
request.extend(compressed_segment)
return bytes(request)
class AsrResponse:
"""ASR响应"""
def __init__(self):
self.code = 0
self.event = 0
self.is_last_package = False
self.payload_sequence = 0
self.payload_size = 0
self.payload_msg = None
def to_dict(self) -> Dict[str, Any]:
return {
"code": self.code,
"event": self.event,
"is_last_package": self.is_last_package,
"payload_sequence": self.payload_sequence,
"payload_size": self.payload_size,
"payload_msg": self.payload_msg
}
class ResponseParser:
"""响应解析器"""
@staticmethod
def parse_response(msg: bytes) -> AsrResponse:
"""解析响应"""
response = AsrResponse()
header_size = msg[0] & 0x0f
message_type = msg[1] >> 4
message_type_specific_flags = msg[1] & 0x0f
serialization_method = msg[2] >> 4
message_compression = msg[2] & 0x0f
payload = msg[header_size*4:]
# 解析message_type_specific_flags
if message_type_specific_flags & 0x01:
response.payload_sequence = struct.unpack('>i', payload[:4])[0]
payload = payload[4:]
if message_type_specific_flags & 0x02:
response.is_last_package = True
if message_type_specific_flags & 0x04:
response.event = struct.unpack('>i', payload[:4])[0]
payload = payload[4:]
# 解析message_type
if message_type == MessageType.SERVER_FULL_RESPONSE:
response.payload_size = struct.unpack('>U', payload[:4])[0]
payload = payload[4:]
elif message_type == MessageType.SERVER_ERROR_RESPONSE:
response.code = struct.unpack('>i', payload[:4])[0]
response.payload_size = struct.unpack('>U', payload[4:8])[0]
payload = payload[8:]
if not payload:
return response
# 解压缩
if message_compression == CompressionType.GZIP:
try:
payload = AudioUtils.gzip_decompress(payload)
except Exception as e:
logger.error(f"Failed to decompress payload: {e}")
return response
# 解析payload
try:
if serialization_method == SerializationType.JSON:
response.payload_msg = json.loads(payload.decode('utf-8'))
except Exception as e:
logger.error(f"Failed to parse payload: {e}")
return response
class SpeechRecognizer:
"""语音识别器"""
def __init__(self, app_key: str = None, access_key: str = None,
url: str = "wss://openspeech.bytedance.com/api/v3/sauc/bigmodel_nostream"):
self.config = AsrConfig(app_key, access_key)
self.url = url
self.seq = 1
async def recognize_file(self, file_path: str) -> List[RecognitionResult]:
"""识别音频文件"""
if not os.path.exists(file_path):
raise FileNotFoundError(f"Audio file not found: {file_path}")
results = []
try:
async with aiohttp.ClientSession() as session:
# 读取音频文件
with open(file_path, 'rb') as f:
content = f.read()
if not AudioUtils.is_wav_file(content):
raise ValueError("Audio file must be in WAV format")
# 获取音频信息
try:
_, _, sample_rate, _, audio_data = AudioUtils.read_wav_info(content)
if sample_rate != DEFAULT_SAMPLE_RATE:
logger.warning(f"Sample rate {sample_rate} != {DEFAULT_SAMPLE_RATE}, may affect recognition accuracy")
except Exception as e:
logger.error(f"Failed to read audio info: {e}")
raise
# 计算分段大小 (200ms per segment)
segment_size = 1 * 2 * DEFAULT_SAMPLE_RATE * 200 // 1000 # channel * bytes_per_sample * sample_rate * duration_ms / 1000
# 创建WebSocket连接
headers = RequestBuilder.new_auth_headers(self.config)
async with session.ws_connect(self.url, headers=headers) as ws:
# 发送完整客户端请求
request = RequestBuilder.new_full_client_request(self.seq)
self.seq += 1
await ws.send_bytes(request)
# 接收初始响应
msg = await ws.receive()
if msg.type == aiohttp.WSMsgType.BINARY:
response = ResponseParser.parse_response(msg.data)
logger.info(f"Initial response: {response.to_dict()}")
# 分段发送音频数据
audio_segments = self._split_audio(audio_data, segment_size)
total_segments = len(audio_segments)
for i, segment in enumerate(audio_segments):
is_last = (i == total_segments - 1)
request = RequestBuilder.new_audio_only_request(
self.seq,
segment,
is_last=is_last
)
await ws.send_bytes(request)
logger.info(f"Sent audio segment {i+1}/{total_segments}")
if not is_last:
self.seq += 1
# 短暂延迟模拟实时流
await asyncio.sleep(0.1)
# 接收识别结果
final_text = ""
while True:
msg = await ws.receive()
if msg.type == aiohttp.WSMsgType.BINARY:
response = ResponseParser.parse_response(msg.data)
if response.payload_msg and 'text' in response.payload_msg:
text = response.payload_msg['text']
if text:
final_text += text
result = RecognitionResult(
text=text,
confidence=0.9, # 默认置信度
is_final=response.is_last_package
)
results.append(result)
logger.info(f"Recognized: {text}")
if response.is_last_package or response.code != 0:
break
elif msg.type == aiohttp.WSMsgType.ERROR:
logger.error(f"WebSocket error: {msg.data}")
break
elif msg.type == aiohttp.WSMsgType.CLOSED:
logger.info("WebSocket connection closed")
break
# 如果没有获得最终结果，创建一个包含所有文本的结果
if final_text and not any(r.is_final for r in results):
final_result = RecognitionResult(
text=final_text,
confidence=0.9,
is_final=True
)
results.append(final_result)
return results
except Exception as e:
logger.error(f"Speech recognition failed: {e}")
raise
def _split_audio(self, data: bytes, segment_size: int) -> List[bytes]:
"""分割音频数据"""
if segment_size <= 0:
return []
segments = []
for i in range(0, len(data), segment_size):
end = i + segment_size
if end > len(data):
end = len(data)
segments.append(data[i:end])
return segments
async def recognize_latest_recording(self, directory: str = ".") -> Optional[RecognitionResult]:
"""识别最新的录音文件"""
# 查找最新的录音文件
recording_files = [f for f in os.listdir(directory) if f.startswith('recording_') and f.endswith('.wav')]
if not recording_files:
logger.warning("No recording files found")
return None
# 按文件名排序（包含时间戳）
recording_files.sort(reverse=True)
latest_file = recording_files[0]
latest_path = os.path.join(directory, latest_file)
logger.info(f"Recognizing latest recording: {latest_file}")
try:
results = await self.recognize_file(latest_path)
if results:
# 返回最终的识别结果
final_results = [r for r in results if r.is_final]
if final_results:
return final_results[-1]
else:
# 如果没有标记为final的结果，返回最后一个
return results[-1]
except Exception as e:
logger.error(f"Failed to recognize latest recording: {e}")
return None
async def main():
"""测试函数"""
import argparse
parser = argparse.ArgumentParser(description="语音识别测试")
parser.add_argument("--file", type=str, help="音频文件路径")
parser.add_argument("--latest", action="store_true", help="识别最新的录音文件")
parser.add_argument("--app-key", type=str, help="SAUC App Key")
parser.add_argument("--access-key", type=str, help="SAUC Access Key")
args = parser.parse_args()
recognizer = SpeechRecognizer(
app_key=args.app_key,
access_key=args.access_key
)
try:
if args.latest:
result = await recognizer.recognize_latest_recording()
if result:
print(f"识别结果: {result.text}")
print(f"置信度: {result.confidence}")
print(f"最终结果: {result.is_final}")
else:
print("未能识别到语音内容")
elif args.file:
results = await recognizer.recognize_file(args.file)
for i, result in enumerate(results):
print(f"结果 {i+1}: {result.text}")
print(f"置信度: {result.confidence}")
print(f"最终结果: {result.is_final}")
print("-" * 40)
else:
print("请指定 --file 或 --latest 参数")
except Exception as e:
print(f"识别失败: {e}")
if __name__ == "__main__":
asyncio.run(main())

344
voice_recorder.py
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@ -1,344 +0,0 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
语音录制模块
基于pyaudio实现，支持语音活动检测(VAD)自动判断录音结束
"""
import pyaudio
import wave
import numpy as np
import time
import os
import threading
from collections import deque
class VoiceRecorder:
"""语音录制器，支持自动检测语音结束"""
def __init__(self,
energy_threshold=500,
silence_threshold=1.0,
min_recording_time=0.5,
max_recording_time=10.0,
sample_rate=16000,
chunk_size=1024,
defer_audio_init=False):
"""
初始化录音器
Args:
energy_threshold: 语音能量阈值
silence_threshold: 静音持续时间阈值（秒）
min_recording_time: 最小录音时间（秒）
max_recording_time: 最大录音时间（秒）
sample_rate: 采样率
chunk_size: 音频块大小
defer_audio_init: 是否延迟音频初始化
"""
self.energy_threshold = energy_threshold
self.silence_threshold = silence_threshold
self.min_recording_time = min_recording_time
self.max_recording_time = max_recording_time
self.sample_rate = sample_rate
self.chunk_size = chunk_size
self.defer_audio_init = defer_audio_init
# 音频参数
self.FORMAT = pyaudio.paInt16
self.CHANNELS = 1
# 状态变量
self.audio = None
self.stream = None
self.recording = False
self.recorded_frames = []
# 语音检测相关
self.silence_start_time = None
self.recording_start_time = None
self.audio_buffer = deque(maxlen=int(sample_rate / chunk_size * 2)) # 2秒缓冲
# 回调函数
self.on_recording_complete = None
self.on_speech_detected = None
if not defer_audio_init:
self._setup_audio()
def _setup_audio(self):
"""设置音频设备"""
try:
self.audio = pyaudio.PyAudio()
# 获取默认输入设备信息
device_info = self.audio.get_default_input_device_info()
print(f"使用音频设备: {device_info['name']}")
except Exception as e:
print(f"音频设备初始化失败: {e}")
raise
def _calculate_energy(self, audio_data):
"""计算音频能量"""
if len(audio_data) == 0:
return 0
# 转换为numpy数组
audio_array = np.frombuffer(audio_data, dtype=np.int16)
# 计算RMS能量
rms = np.sqrt(np.mean(audio_array ** 2))
return rms
def _is_speech(self, audio_data):
"""判断是否为语音"""
energy = self._calculate_energy(audio_data)
return energy > self.energy_threshold
def _open_stream(self):
"""打开音频流"""
if self.stream is not None:
return
self.stream = self.audio.open(
format=self.FORMAT,
channels=self.CHANNELS,
rate=self.sample_rate,
input=True,
frames_per_buffer=self.chunk_size
)
def _close_stream(self):
"""关闭音频流"""
if self.stream:
self.stream.stop_stream()
self.stream.close()
self.stream = None
def start_listening(self):
"""开始监听语音"""
if self.recording:
print("正在录音中...")
return
self._open_stream()
self.recording = True
self.recorded_frames = []
self.silence_start_time = None
self.recording_start_time = None
print("开始监听语音...")
# 在新线程中录音
recording_thread = threading.Thread(target=self._record_loop)
recording_thread.daemon = True
recording_thread.start()
def _record_loop(self):
"""录音循环"""
try:
while self.recording:
# 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
energy = self._calculate_energy(data)
# 添加到缓冲区
self.audio_buffer.append(data)
# 检测语音活动
if energy > self.energy_threshold:
# 检测到语音
if self.recording_start_time is None:
# 开始录音
self.recording_start_time = time.time()
self.silence_start_time = None
self.recorded_frames = list(self.audio_buffer) # 包含之前的音频
print("🎤 检测到语音，开始录音...")
if self.on_speech_detected:
self.on_speech_detected()
# 重置静音计时
self.silence_start_time = None
# 录音
self.recorded_frames.append(data)
elif self.recording_start_time is not None:
# 之前有语音，现在检查是否静音
if self.silence_start_time is None:
self.silence_start_time = time.time()
# 继续录音
self.recorded_frames.append(data)
# 检查是否静音超时
silence_duration = time.time() - self.silence_start_time
if silence_duration > self.silence_threshold:
recording_duration = time.time() - self.recording_start_time
# 检查最小录音时间
if recording_duration >= self.min_recording_time:
print(f"静音 {silence_duration:.1f}s，结束录音")
self.stop_recording()
break
else:
print(f"录音时间太短 ({recording_duration:.1f}s)，继续等待...")
self.silence_start_time = time.time()
# 检查最大录音时间
if self.recording_start_time is not None:
recording_duration = time.time() - self.recording_start_time
if recording_duration > self.max_recording_time:
print(f"达到最大录音时间 {self.max_recording_time}s，结束录音")
self.stop_recording()
break
# 短暂休眠
time.sleep(0.01)
except Exception as e:
print(f"录音过程中发生错误: {e}")
self.stop_recording()
def stop_recording(self):
"""停止录音"""
if not self.recording:
return
self.recording = False
self._close_stream()
if len(self.recorded_frames) > 0:
# 保存录音
audio_data = b''.join(self.recorded_frames)
print(f"录音完成，共 {len(self.recorded_frames)} 帧")
print(f"录音时长: {len(audio_data) / (self.sample_rate * 2):.2f} 秒")
# 调用回调函数
if self.on_recording_complete:
self.on_recording_complete(audio_data)
# 重置状态
self.recorded_frames = []
self.silence_start_time = None
self.recording_start_time = None
def save_audio(self, audio_data, filename):
"""保存音频到文件"""
try:
with wave.open(filename, 'wb') as wf:
wf.setnchannels(self.CHANNELS)
wf.setsampwidth(self.audio.get_sample_size(self.FORMAT))
wf.setframerate(self.sample_rate)
wf.writeframes(audio_data)
print(f"音频已保存到: {filename}")
return True
except Exception as e:
print(f"保存音频失败: {e}")
return False
def set_recording_complete_callback(self, callback):
"""设置录音完成回调函数"""
self.on_recording_complete = callback
def set_speech_detected_callback(self, callback):
"""设置语音检测回调函数"""
self.on_speech_detected = callback
def adjust_sensitivity(self, energy_threshold=None, silence_threshold=None):
"""调整灵敏度"""
if energy_threshold is not None:
self.energy_threshold = energy_threshold
print(f"能量阈值调整为: {energy_threshold}")
if silence_threshold is not None:
self.silence_threshold = silence_threshold
print(f"静音阈值调整为: {silence_threshold}秒")
def get_audio_level(self):
"""获取当前音频级别"""
if len(self.audio_buffer) > 0:
latest_data = self.audio_buffer[-1]
return self._calculate_energy(latest_data)
return 0
def cleanup(self):
"""清理资源"""
self.stop_recording()
if self.audio:
self.audio.terminate()
self.audio = None
def main():
"""测试录音功能"""
print("🎙️ 语音录制测试")
print("=" * 50)
print("配置:")
print("- 能量阈值: 500")
print("- 静音阈值: 1.0秒")
print("- 最小录音时间: 0.5秒")
print("- 最大录音时间: 10秒")
print("=" * 50)
print("请说话测试录音功能...")
print("按 Ctrl+C 退出")
def on_recording_complete(audio_data):
"""录音完成回调"""
# 保存录音文件
timestamp = time.strftime("%Y%m%d_%H%M%S")
filename = f"recording_{timestamp}.wav"
recorder.save_audio(audio_data, filename)
print(f"✅ 录音文件已保存: {filename}")
# 显示录音信息
duration = len(audio_data) / (recorder.sample_rate * 2)
print(f"录音时长: {duration:.2f} 秒")
def on_speech_detected():
"""检测到语音回调"""
print("🔊 检测到语音活动...")
# 创建录音器
recorder = VoiceRecorder(
energy_threshold=500,
silence_threshold=1.0,
min_recording_time=0.5,
max_recording_time=10.0
)
# 设置回调
recorder.set_recording_complete_callback(on_recording_complete)
recorder.set_speech_detected_callback(on_speech_detected)
try:
# 开始监听
recorder.start_listening()
# 保持程序运行
while True:
time.sleep(0.1)
# 显示当前音频级别（可选）
level = recorder.get_audio_level()
if level > 100:
print(f"当前音频级别: {level:.0f}", end='\r')
except KeyboardInterrupt:
print("\n👋 退出录音测试")
finally:
recorder.cleanup()
if __name__ == "__main__":
main()