whisper
Provides OpenAI's Whisper multilingual speech recognition model for speech-to-text, podcast/video transcription, and meeting minutes across 99 languages.
npx skills add davila7/claude-code-templates --skill whisperBefore / After Comparison
1 组Using traditional speech recognition tools, transcription accuracy is low in multilingual or noisy environments, it's time-consuming, requires extensive manual proofreading, and is costly.
By adopting the OpenAI Whisper model, high-accuracy multilingual speech-to-text is achieved, significantly reducing manual intervention and proofreading time, and lowering costs.
description SKILL.md
whisper
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
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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+
-
Model Card: https://github.com/openai/whisper/blob/main/model-card.md
-
Colab: Available in repo
-
License: MIT
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