Agent skill
Automatic Speech Recognition (ASR)
Transcribe audio segments to text using Whisper models. Use larger models (small, base, medium, large-v3) for better accuracy, or faster-whisper for optimized performance. Always align transcription timestamps with diarization segments for accurate speaker-labeled subtitles.
Install this agent skill to your Project
npx add-skill https://github.com/benchflow-ai/skillsbench/tree/main/tasks/speaker-diarization-subtitles/environment/skills/automatic-speech-recognition
SKILL.md
Automatic Speech Recognition (ASR)
Overview
After speaker diarization, you need to transcribe each speech segment to text. Whisper is the current state-of-the-art for ASR, with multiple model sizes offering different trade-offs between accuracy and speed.
When to Use
- After speaker diarization is complete
- Need to generate speaker-labeled transcripts
- Creating subtitles from audio segments
- Converting speech segments to text
Whisper Model Selection
Model Size Comparison
| Model | Size | Speed | Accuracy | Best For |
|---|---|---|---|---|
| tiny | 39M | Fastest | Lowest | Quick testing, low accuracy needs |
| base | 74M | Fast | Low | Fast processing with moderate accuracy |
| small | 244M | Medium | Good | Recommended balance |
| medium | 769M | Slow | Very Good | High accuracy needs |
| large-v3 | 1550M | Slowest | Best | Maximum accuracy |
Recommended: Use small or large-v3
For best accuracy (recommended for this task):
import whisper
model = whisper.load_model("large-v3") # Best accuracy
result = model.transcribe(audio_path)
For balanced performance:
import whisper
model = whisper.load_model("small") # Good balance
result = model.transcribe(audio_path)
Faster-Whisper (Optimized Alternative)
For faster processing with similar accuracy, use faster-whisper:
from faster_whisper import WhisperModel
# Use small model with CPU int8 quantization
model = WhisperModel("small", device="cpu", compute_type="int8")
# Transcribe
segments, info = model.transcribe(audio_path, beam_size=5)
# Process segments
for segment in segments:
print(f"[{segment.start:.2f}s -> {segment.end:.2f}s] {segment.text}")
Advantages:
- Faster than standard Whisper
- Lower memory usage with quantization
- Similar accuracy to standard Whisper
Aligning Transcriptions with Diarization Segments
After diarization, you need to map Whisper transcriptions to speaker segments:
# After diarization, you have turns with speaker labels
turns = [
{'start': 0.8, 'duration': 0.86, 'speaker': 'SPEAKER_01'},
{'start': 5.34, 'duration': 0.21, 'speaker': 'SPEAKER_01'},
# ...
]
# Run Whisper transcription
model = whisper.load_model("large-v3")
result = model.transcribe(audio_path)
# Map transcriptions to turns
transcripts = {}
for i, turn in enumerate(turns):
turn_start = turn['start']
turn_end = turn['start'] + turn['duration']
# Find overlapping Whisper segments
overlapping_text = []
for seg in result['segments']:
seg_start = seg['start']
seg_end = seg['end']
# Check if Whisper segment overlaps with diarization turn
if seg_start < turn_end and seg_end > turn_start:
overlapping_text.append(seg['text'].strip())
# Combine overlapping segments
transcripts[i] = ' '.join(overlapping_text) if overlapping_text else '[INAUDIBLE]'
Handling Empty or Inaudible Segments
# If no transcription found for a segment
if not overlapping_text:
transcripts[i] = '[INAUDIBLE]'
# Or skip very short segments
if turn['duration'] < 0.3:
transcripts[i] = '[INAUDIBLE]'
Language Detection
Whisper can auto-detect language, but you can also specify:
# Auto-detect (recommended)
result = model.transcribe(audio_path)
# Or specify language for better accuracy
result = model.transcribe(audio_path, language="en")
Best Practices
- Use larger models for better accuracy:
smallminimum,large-v3for best results - Align timestamps carefully: Match Whisper segments with diarization turns
- Handle overlaps: Multiple Whisper segments may overlap with one diarization turn
- Handle gaps: Some diarization turns may have no corresponding transcription
- Post-process text: Clean up punctuation, capitalization if needed
Common Issues
- Low transcription accuracy: Use larger model (small → medium → large-v3)
- Slow processing: Use faster-whisper or smaller model
- Misaligned timestamps: Check time alignment between diarization and transcription
- Missing transcriptions: Check for very short segments or silence
Integration with Subtitle Generation
After transcription, combine with speaker labels for subtitles:
def generate_subtitles_ass(turns, transcripts, output_path):
# ... header code ...
for i, turn in enumerate(turns):
start_time = format_time(turn['start'])
end_time = format_time(turn['start'] + turn['duration'])
speaker = turn['speaker']
text = transcripts.get(i, "[INAUDIBLE]")
# Format: SPEAKER_XX: text
f.write(f"Dialogue: 0,{start_time},{end_time},Default,,0,0,0,,{speaker}: {text}\n")
Performance Tips
- For accuracy: Use
large-v3model - For speed: Use
faster-whisperwithsmallmodel - For memory: Use
faster-whisperwithint8quantization - Batch processing: Process multiple segments together if possible
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