whisper

Multilingual speech recognition with 99 languages, transcription, translation, and six model sizes from 39M to 1550M parameters. Supports transcription, translation to English, language identification, and word-level timestamps across 99 languages trained on 680,000 hours of audio Six model sizes (tiny through large) with configurable speed/quality tradeoffs; turbo model offers 8× speedup over large with comparable quality GPU acceleration delivers 10–20× faster processing; CPU transcription available for resource-constrained environments Command-line and Python API interfaces; outputs to plain text, SRT subtitles, WebVTT, or JSON with segment timing Initial prompt support improves accuracy on technical terms and domain-specific vocabulary; temperature fallback for low-confidence segments

INSTALLATION
npx skills add https://github.com/davila7/claude-code-templates --skill whisper
Run in your project or agent environment. Adjust flags if your CLI version differs.

SKILL.md

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

  • Colab: Available in repo
  • License: MIT
BrowserAct

Let your agent run on any real-world website

Bypass CAPTCHA & anti-bot for free. Start local, scale to cloud.

Explore BrowserAct Skills →

Stop writing automation&scrapers

Install the CLI. Run your first Skill in 30 seconds. Scale when you're ready.

Start free
free · no credit card