faster whisper从多媒体语音材料中抽取出文本-2
为脚本添加每个音频的时长统计和每个音频转换所有的耗时统计
安装依赖
pip install faster-whisper pydub
"""
批量转录当前目录下的 .mp3 文件,使用 faster-whisper
新增功能:
- 每个音频的时长(秒)
- 每个音频的转录耗时(秒)
- 总计统计:总音频时长、总转录耗时、平均实时倍率
"""
import os
import sys
import time
from pathlib import Path
from typing import List, Tuple
from faster_whisper import WhisperModel
from pydub import AudioSegment
# ================== 配置区 ==================
MODEL_SIZE = "small" # 可选: tiny, base, small, medium, large
DEVICE = "cpu" # cpu 或 cuda
COMPUTE_TYPE = "int8" # int8, float16, float32 (CPU 推荐 int8)
VAD_FILTER = True # 启用语音活动检测,去除静音
OUTPUT_FORMAT = "txt" # 只输出 .txt
VERBOSE = True # 是否显示详细日志
# ===========================================
def get_audio_duration(audio_path: Path) -> float:
"""使用 pydub 获取音频时长(秒)"""
try:
audio = AudioSegment.from_file(str(audio_path))
return len(audio) / 1000.0 # 毫秒 → 秒
except Exception as e:
print(f"无法获取 {audio_path.name} 时长: {e}", file=sys.stderr)
return 0.0
def transcribe_audio(
audio_path: Path, model: WhisperModel
) -> Tuple[str, float, float]:
"""
转录单个音频文件
返回: (文本内容, 音频时长秒, 转录耗时秒)
"""
duration = get_audio_duration(audio_path)
print(f"转录: {audio_path.name} ({duration:.2f}s) → {audio_path.stem}.txt")
start_time = time.perf_counter()
segments, info = model.transcribe(
str(audio_path),
language=None, # 自动检测
beam_size=5,
vad_filter=VAD_FILTER,
vad_parameters=dict(min_silence_duration_ms=500),
word_timestamps=False,
)
elapsed = time.perf_counter() - start_time
text_lines = []
for segment in segments:
line = segment.text.strip()
text_lines.append(line)
if VERBOSE:
print(f"[{segment.start:06.2f}s --> {segment.end:06.2f}s] {line}", flush=True)
return "n".join(text_lines), duration, elapsed
def format_time(seconds: float) -> str:
"""将秒数格式化为 h:mm:ss"""
hours = int(seconds // 3600)
minutes = int((seconds % 3600) // 60)
secs = seconds % 60
return f"{hours}:{minutes:02d}:{secs:05.2f}"
def main():
print("=== faster-whisper 批量转录(带时长与耗时统计)===")
current_dir = Path(".")
mp3_files = sorted(current_dir.glob("*.mp3"))
if not mp3_files:
print("未找到 .mp3 文件,退出。")
return
# 加载模型(只加载一次)
print(f"正在加载模型 {MODEL_SIZE} ({DEVICE}, {COMPUTE_TYPE})...")
model = WhisperModel(MODEL_SIZE, device=DEVICE, compute_type=COMPUTE_TYPE)
processed = 0
total_audio_duration = 0.0
total_transcribe_time = 0.0
results = []
for mp3_path in mp3_files:
txt_path = mp3_path.with_suffix(".txt")
if txt_path.exists():
duration = get_audio_duration(mp3_path)
print(f"跳过: {txt_path.name} 已存在 ({duration:.2f}s)")
total_audio_duration += duration
continue
try:
text, duration, elapsed = transcribe_audio(mp3_path, model)
txt_path.write_text(text, encoding="utf-8")
total_audio_duration += duration
total_transcribe_time += elapsed
processed += 1
rtf = elapsed / duration if duration > 0 else float('inf')
print(f"完成: {mp3_path.name} | 时长 {duration:.2f}s | 耗时 {elapsed:.2f}s | RTF {rtf:.2f}x")
results.append((mp3_path.name, duration, elapsed, rtf))
except Exception as e:
print(f"错误转录 {mp3_path.name}: {e}", file=sys.stderr)
# ================== 汇总统计 ==================
print("n" + "=" * 60)
print("转录完成汇总")
print("=" * 60)
print(f"成功处理文件数 : {processed}")
print(f"总音频时长 : {format_time(total_audio_duration)}")
print(f"总转录耗时 : {format_time(total_transcribe_time)}")
if total_audio_duration > 0:
avg_rtf = total_transcribe_time / total_audio_duration
print(f"平均实时倍率(RTF): {avg_rtf:.2f}x")
else:
print(f"平均实时倍率(RTF): N/A")
if results:
print(f"n明细列表:")
print(f"{'文件名':<40} {'音频时长':>10} {'转录耗时':>10} {'RTF':>8}")
print("-" * 70)
for name, dur, ela, rtf in results:
print(f"{name:<40} {dur:10.2f}s {ela:10.2f}s {rtf:8.2f}x")
print("=" * 60)
if __name__ == "__main__":
main()