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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
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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 📦 Ships with claude-code-templates
</> SKILL.md
multimodal-whisper.SKILL.md
---
name : whisper
description : 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.
version : 1.0.0
author : Orchestra Research
license : MIT
tags : [Whisper, Speech Recognition, ASR, Multimodal, Multilingual, OpenAI, Speech-To-Text, Transcription, Translation, Audio Processing]
dependencies : [openai-whisper, transformers, torch]
---
# 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
Flowy AI Flows that just work. Hand-picked Claude Code plugins that fire the right skill as you prompt.
**Use alternatives instead**:
- **AssemblyAI**: Managed API, speaker diarization
- **Deepgram**: Real-time streaming ASR
- **Google Speech-to-Text**: Cloud-based
## Quick start
### Installation
```bash
# 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
```python
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
```python
# 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
```python
# 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
```python
# 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
```python
# 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
```python
# 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
```python
# 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
```bash
# 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
```python
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
```python
# 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
```python
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
```bash
# Generate SRT subtitles
whisper video.mp4 --output_format srt --language English
# Output: video.srt
```
### With LangChain
```python
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
```bash
# Use ffmpeg to extract audio
ffmpeg -i video.mp4 -vn -acodec pcm_s16le audio.wav
# Then transcribe
whisper audio.wav
```
## Best practices
1. **Use turbo model** - Best speed/quality for English
2. **Specify language** - Faster than auto-detect
3. **Add initial prompt** - Improves technical terms
4. **Use GPU** - 10-20× faster
5. **Batch process** - More efficient
6. **Convert to WAV** - Better compatibility
7. **Split long audio** - <30 min chunks
8. **Check language support** - Quality varies by language
9. **Use faster-whisper** - 4× faster than openai-whisper
10. **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
1. **Hallucinations** - May repeat or invent text
2. **Long-form accuracy** - Degrades on >30 min audio
3. **Speaker identification** - No diarization
4. **Accents** - Quality varies
5. **Background noise** - Can affect accuracy
6. **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
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