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import sys
import numpy as np
import soundfile as sf
from typing import List, Tuple, Optional, Dict
import webrtcvad
from dataclasses import dataclass, asdict
from scipy import signal
import json
import os
from datetime import datetime
@dataclass
class AudioSegment:
start_time: float # 开始时间(秒)
end_time: float # 结束时间(秒)
audio_data: np.ndarray # 音频数据
is_speech: bool # 是否包含语音
class AudioProcessor:
def __init__(self,
sample_rate: int = 16000,
frame_duration_ms: int = 20,
vad_level: int = 0, # 降低VAD灵敏度
min_silence_duration: float = 0.3, # 静音持续时间
min_speech_duration: float = 0.3, # 增加最小语音持续时间,确保完整句子
amplitude_threshold: float = 0.0015): # 进一步降低振幅阈值
"""
初始化音频处理器
Args:
sample_rate: 采样率
frame_duration_ms: VAD帧长度(毫秒)
vad_level: VAD灵敏度 (0-3)
min_silence_duration: 最小静音持续时间(秒)
min_speech_duration: 最小语音片段长度(秒)
amplitude_threshold: 振幅阈值
"""
self.sample_rate = sample_rate
self.frame_duration_ms = frame_duration_ms
self.frame_size = int(sample_rate * frame_duration_ms / 1000)
self.vad = webrtcvad.Vad(vad_level)
self.min_silence_frames = int(min_silence_duration * 1000 / frame_duration_ms)
self.min_speech_frames = int(min_speech_duration * 1000 / frame_duration_ms)
self.amplitude_threshold = amplitude_threshold
def _is_speech_frame(self, frame: np.ndarray) -> bool:
"""
判断一帧是否包含语音
"""
# 确保帧长度正确
if len(frame) != self.frame_size:
return False
# 将float32转换为int16,并确保值在范围内
frame_int16 = np.clip(frame * 32768, -32768, 32767).astype(np.int16)
# 使用振幅判断
frame_amplitude = np.max(np.abs(frame))
if frame_amplitude < self.amplitude_threshold:
return False
# 使用VAD判断
try:
return self.vad.is_speech(frame_int16.tobytes(), self.sample_rate)
except Exception as e:
print(f"VAD处理出错: {e}")
# 如果VAD失败,仅使用振幅判断
return frame_amplitude >= self.amplitude_threshold * 2
def process_audio_file(self, audio_path: str) -> List[AudioSegment]:
"""
处理音频文件,返回切割后的片段列表
"""
# 读取音频文件
print(f"正在读取音频文件: {audio_path}")
audio_data, sample_rate = sf.read(audio_path)
print(f"音频采样率: {sample_rate}Hz, 形状: {audio_data.shape}")
if sample_rate != self.sample_rate:
print(f"正在重采样音频从 {sample_rate}Hz 到 {self.sample_rate}Hz")
# 使用scipy的resample函数进行重采样
num_samples = int(len(audio_data) * self.sample_rate / sample_rate)
audio_data = signal.resample(audio_data, num_samples)
print(f"重采样后音频长度: {len(audio_data)} 采样点")
if len(audio_data.shape) > 1:
print("检测到多声道音频,正在转换为单声道")
audio_data = audio_data.mean(axis=1) # 转换为单声道
# 初始化结果列表
segments: List[AudioSegment] = []
print(f"开始处理音频,总长度: {len(audio_data)} 采样点 ({len(audio_data)/self.sample_rate:.2f}秒)")
# 当前处理的状态
current_segment_start = 0
silence_frame_count = 0
is_in_speech = False
# 按帧处理音频
total_frames = len(audio_data) // self.frame_size
speech_frames = 0
for i in range(0, len(audio_data), self.frame_size):
# 确保帧长度正确
frame = audio_data[i:i + self.frame_size]
if len(frame) < self.frame_size:
# 对于最后一个不完整帧,补零处理
frame = np.pad(frame, (0, self.frame_size - len(frame)), 'constant')
is_speech = self._is_speech_frame(frame)
if is_speech:
speech_frames += 1
if is_speech and not is_in_speech:
# 开始新的语音段
current_segment_start = i
is_in_speech = True
silence_frame_count = 0
print(f"\n检测到语音开始,位置: {i/self.sample_rate:.2f}秒")
elif not is_speech and is_in_speech:
silence_frame_count += 1
# 如果静音持续足够长,结束当前语音段
if silence_frame_count >= self.min_silence_frames:
segment_end = i - (silence_frame_count * self.frame_size)
duration_frames = (segment_end - current_segment_start) // self.frame_size
# 只保存超过最小长度的片段
if duration_frames >= self.min_speech_frames:
start_time = current_segment_start / self.sample_rate
end_time = segment_end / self.sample_rate
print(f"保存语音片段: {start_time:.2f}s -> {end_time:.2f}s (持续时间: {end_time-start_time:.2f}s)")
segments.append(AudioSegment(
start_time=start_time,
end_time=end_time,
audio_data=audio_data[current_segment_start:segment_end],
is_speech=True
))
else:
print(f"丢弃过短的语音片段: {duration_frames * self.frame_duration_ms / 1000:.2f}s")
is_in_speech = False
# 处理最后一个语音段
if is_in_speech:
segment_end = len(audio_data)
duration_frames = (segment_end - current_segment_start) // self.frame_size
if duration_frames >= self.min_speech_frames:
start_time = current_segment_start / self.sample_rate
end_time = segment_end / self.sample_rate
print(f"保存最后的语音片段: {start_time:.2f}s -> {end_time:.2f}s (持续时间: {end_time-start_time:.2f}s)")
segments.append(AudioSegment(
start_time=start_time,
end_time=end_time,
audio_data=audio_data[current_segment_start:segment_end],
is_speech=True
))
else:
print(f"丢弃过短的最后语音片段: {duration_frames * self.frame_duration_ms / 1000:.2f}s")
print(f"\n音频处理完成:")
print(f"总帧数: {total_frames}")
print(f"语音帧数: {speech_frames}")
print(f"检测到的语音片段数: {len(segments)}")
# print("中断程序以进行调试")
# sys.exit(1)
# 保存中间结果到临时文件
temp_dir = "../dataset/audio/temp"
os.makedirs(temp_dir, exist_ok=True)
# 准备保存的数据
temp_data = {
"audio_file": audio_path,
"timestamp": datetime.now().strftime("%Y%m%d_%H%M%S"),
"total_frames": total_frames,
"speech_frames": speech_frames,
"segments": [
{
"start_time": seg.start_time,
"end_time": seg.end_time,
"duration": seg.end_time - seg.start_time,
"is_speech": seg.is_speech
}
for seg in segments
]
}
# 保存临时结果
base_name = os.path.splitext(os.path.basename(audio_path))[0]
temp_path = os.path.join(temp_dir, f"{base_name}_segments_{temp_data['timestamp']}.json")
with open(temp_path, 'w', encoding='utf-8') as f:
json.dump(temp_data, f, ensure_ascii=False, indent=2)
print(f"\n临时结果已保存到: {temp_path}")
return segments
def save_segment(self, segment: AudioSegment, output_path: str):
"""
保存音频片段到文件
"""
sf.write(output_path, segment.audio_data, self.sample_rate)
if __name__ == "__main__":
# 测试代码
processor = AudioProcessor()
# 示例:处理一个音频文件
audio_path = "dataset/audio/test.wav" # 替换为实际的音频文件路径
try:
segments = processor.process_audio_file(audio_path)
print(f"检测到 {len(segments)} 个语音片段:")
for i, segment in enumerate(segments):
print(f"片段 {i+1}: {segment.start_time:.2f}s -> {segment.end_time:.2f}s")
except Exception as e:
print(f"处理音频时出错: {e}")
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