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A library for transcribing audio files using Whisper models

Project description

Whisper Transcriber

A Python library for transcribing audio files using Whisper models with intelligent silence detection and segmentation.

Installation

pip install whisper-transcriber

Requirements

  • Python 3.7 or higher
  • ffmpeg and ffprobe installed on your system

Features

  • Intelligent silence detection for natural segmentation
  • Adaptive audio analysis for optimal threshold detection
  • High-quality transcription using Whisper models
  • Support for various audio formats
  • SRT subtitle output

Usage

Command Line

# Basic usage
whisper-transcribe audio_file.mp3

# Advanced usage
whisper-transcribe audio_file.mp3 -o transcript.srt -m openai/whisper-small \
  --min-segment 5 \
  --max-segment 15 \
  --silence-duration 0.2 \
  --sample-rate 16000 \
  --batch-size 8 \
  --normalize \
  --hf-token YOUR_HF_TOKEN \
  --no-transcripts \
  --no-timestamps

Available Arguments:

  • input: Input audio file or directory (required)
  • -o, --output: Output file path (default: input filename with .srt extension)
  • -m, --model: Whisper model to use (default: openai/whisper-small)
  • --hf-token: HuggingFace API token
  • --min-segment: Minimum segment length in seconds (default: 5)
  • --max-segment: Maximum segment length in seconds (default: 15)
  • --silence-duration: Minimum silence duration in seconds (default: 0.2)
  • --sample-rate: Audio sample rate (default: 16000)
  • --batch-size: Batch size for transcription (default: 8)
  • --normalize: Normalize audio volume
  • --no-text-normalize: Skip text normalization
  • --no-transcripts: Don't print transcripts during processing
  • --no-timestamps: Don't print timestamps during processing

Python Library

from whisper_transcriber import WhisperTranscriber

# Initialize the transcriber
transcriber = WhisperTranscriber(model_name="openai/whisper-small", hf_token="YOUR_HF_TOKEN")

# Transcribe an audio file
results = transcriber.transcribe(
    "audio_file.mp3",
    output="transcript.srt",
    min_segment=5,
    max_segment=15,
    silence_duration=0.2,
    sample_rate=16000,
    batch_size=8,
    normalize=True,
    normalize_text=True,
    print_transcripts=True,
    print_timestamps=True
)

# Access the transcription results
for i, segment in enumerate(results):
    print(f"Segment {i+1}: {segment['transcript']}")

License

MIT

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