Claude-code-templates whisper
OpenAI's general-purpose speech recognition model. Supports 99 languages, transcription, translation to English, and language identification. Six model sizes from tiny (39M params) to large (1550M params). Use for speech-to-text, podcast transcription, or multilingual audio processing. Best for robust, multilingual ASR.
install
source · Clone the upstream repo
git clone https://github.com/davila7/claude-code-templates
Claude Code · Install into ~/.claude/skills/
T=$(mktemp -d) && git clone --depth=1 https://github.com/davila7/claude-code-templates "$T" && mkdir -p ~/.claude/skills && cp -r "$T/cli-tool/components/skills/ai-research/multimodal-whisper" ~/.claude/skills/davila7-claude-code-templates-whisper && rm -rf "$T"
manifest:
cli-tool/components/skills/ai-research/multimodal-whisper/SKILL.mdsource content
Whisper - Robust Speech Recognition
OpenAI's multilingual speech recognition model.
When to use Whisper
Use when:
- Speech-to-text transcription (99 languages)
- Podcast/video transcription
- Meeting notes automation
- Translation to English
- Noisy audio transcription
- Multilingual audio processing
Metrics:
- 72,900+ GitHub stars
- 99 languages supported
- Trained on 680,000 hours of audio
- MIT License
Use alternatives instead:
- AssemblyAI: Managed API, speaker diarization
- Deepgram: Real-time streaming ASR
- Google Speech-to-Text: Cloud-based
Quick start
Installation
# Requires Python 3.8-3.11 pip install -U openai-whisper # Requires ffmpeg # macOS: brew install ffmpeg # Ubuntu: sudo apt install ffmpeg # Windows: choco install ffmpeg
Basic transcription
import whisper # Load model model = whisper.load_model("base") # Transcribe result = model.transcribe("audio.mp3") # Print text print(result["text"]) # Access segments for segment in result["segments"]: print(f"[{segment['start']:.2f}s - {segment['end']:.2f}s] {segment['text']}")
Model sizes
# Available models models = ["tiny", "base", "small", "medium", "large", "turbo"] # Load specific model model = whisper.load_model("turbo") # Fastest, good quality
| Model | Parameters | English-only | Multilingual | Speed | VRAM |
|---|---|---|---|---|---|
| tiny | 39M | ✓ | ✓ | ~32x | ~1 GB |
| base | 74M | ✓ | ✓ | ~16x | ~1 GB |
| small | 244M | ✓ | ✓ | ~6x | ~2 GB |
| medium | 769M | ✓ | ✓ | ~2x | ~5 GB |
| large | 1550M | ✗ | ✓ | 1x | ~10 GB |
| turbo | 809M | ✗ | ✓ | ~8x | ~6 GB |
Recommendation: Use
turbo for best speed/quality, base for prototyping
Transcription options
Language specification
# Auto-detect language result = model.transcribe("audio.mp3") # Specify language (faster) result = model.transcribe("audio.mp3", language="en") # Supported: en, es, fr, de, it, pt, ru, ja, ko, zh, and 89 more
Task selection
# Transcription (default) result = model.transcribe("audio.mp3", task="transcribe") # Translation to English result = model.transcribe("spanish.mp3", task="translate") # Input: Spanish audio → Output: English text
Initial prompt
# Improve accuracy with context result = model.transcribe( "audio.mp3", initial_prompt="This is a technical podcast about machine learning and AI." ) # Helps with: # - Technical terms # - Proper nouns # - Domain-specific vocabulary
Timestamps
# Word-level timestamps result = model.transcribe("audio.mp3", word_timestamps=True) for segment in result["segments"]: for word in segment["words"]: print(f"{word['word']} ({word['start']:.2f}s - {word['end']:.2f}s)")
Temperature fallback
# Retry with different temperatures if confidence low result = model.transcribe( "audio.mp3", temperature=(0.0, 0.2, 0.4, 0.6, 0.8, 1.0) )
Command line usage
# Basic transcription whisper audio.mp3 # Specify model whisper audio.mp3 --model turbo # Output formats whisper audio.mp3 --output_format txt # Plain text whisper audio.mp3 --output_format srt # Subtitles whisper audio.mp3 --output_format vtt # WebVTT whisper audio.mp3 --output_format json # JSON with timestamps # Language whisper audio.mp3 --language Spanish # Translation whisper spanish.mp3 --task translate
Batch processing
import os audio_files = ["file1.mp3", "file2.mp3", "file3.mp3"] for audio_file in audio_files: print(f"Transcribing {audio_file}...") result = model.transcribe(audio_file) # Save to file output_file = audio_file.replace(".mp3", ".txt") with open(output_file, "w") as f: f.write(result["text"])
Real-time transcription
# For streaming audio, use faster-whisper # pip install faster-whisper from faster_whisper import WhisperModel model = WhisperModel("base", device="cuda", compute_type="float16") # Transcribe with streaming segments, info = model.transcribe("audio.mp3", beam_size=5) for segment in segments: print(f"[{segment.start:.2f}s -> {segment.end:.2f}s] {segment.text}")
GPU acceleration
import whisper # Automatically uses GPU if available model = whisper.load_model("turbo") # Force CPU model = whisper.load_model("turbo", device="cpu") # Force GPU model = whisper.load_model("turbo", device="cuda") # 10-20× faster on GPU
Integration with other tools
Subtitle generation
# Generate SRT subtitles whisper video.mp4 --output_format srt --language English # Output: video.srt
With LangChain
from langchain.document_loaders import WhisperTranscriptionLoader loader = WhisperTranscriptionLoader(file_path="audio.mp3") docs = loader.load() # Use transcription in RAG from langchain_chroma import Chroma from langchain_openai import OpenAIEmbeddings vectorstore = Chroma.from_documents(docs, OpenAIEmbeddings())
Extract audio from video
# Use ffmpeg to extract audio ffmpeg -i video.mp4 -vn -acodec pcm_s16le audio.wav # Then transcribe whisper audio.wav
Best practices
- Use turbo model - Best speed/quality for English
- Specify language - Faster than auto-detect
- Add initial prompt - Improves technical terms
- Use GPU - 10-20× faster
- Batch process - More efficient
- Convert to WAV - Better compatibility
- Split long audio - <30 min chunks
- Check language support - Quality varies by language
- Use faster-whisper - 4× faster than openai-whisper
- Monitor VRAM - Scale model size to hardware
Performance
| Model | Real-time factor (CPU) | Real-time factor (GPU) |
|---|---|---|
| tiny | ~0.32 | ~0.01 |
| base | ~0.16 | ~0.01 |
| turbo | ~0.08 | ~0.01 |
| large | ~1.0 | ~0.05 |
Real-time factor: 0.1 = 10× faster than real-time
Language support
Top-supported languages:
- English (en)
- Spanish (es)
- French (fr)
- German (de)
- Italian (it)
- Portuguese (pt)
- Russian (ru)
- Japanese (ja)
- Korean (ko)
- Chinese (zh)
Full list: 99 languages total
Limitations
- Hallucinations - May repeat or invent text
- Long-form accuracy - Degrades on >30 min audio
- Speaker identification - No diarization
- Accents - Quality varies
- Background noise - Can affect accuracy
- Real-time latency - Not suitable for live captioning
Resources
- GitHub: https://github.com/openai/whisper ⭐ 72,900+
- Paper: https://arxiv.org/abs/2212.04356
- Model Card: https://github.com/openai/whisper/blob/main/model-card.md
- Colab: Available in repo
- License: MIT