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Whisper is a general-purpose speech recognition model. It is trained on a large dataset of diverse audio and is also a multitasking model that can perform multilingual speech recognition, speech translation, and language identification.
A Transformer sequence-to-sequence model is trained on various speech processing tasks, including multilingual speech recognition, speech translation, spoken language identification, and voice activity detection. These tasks are jointly represented as a sequence of tokens to be predicted by the decoder, allowing a single model to replace many stages of a traditional speech-processing pipeline. The multitask training format uses a set of special tokens that serve as task specifiers or classification targets.
We used Python 3.9.9 and PyTorch 1.10.1 to train and test our models, but the codebase is expected to be compatible with Python 3.8-3.11 and recent PyTorch versions. The codebase also depends on a few Python packages, most notably OpenAI's tiktoken for their fast tokenizer implementation. You can download and install (or update to) the latest release of Whisper with the following command:
pip install -U openai-whisper
Alternatively, the following command will pull and install the latest commit from this repository, along with its Python dependencies:
pip install git+https://github.com/openai/whisper.git
To update the package to the latest version of this repository, please run:
pip install --upgrade --no-deps --force-reinstall git+https://github.com/openai/whisper.git
It also requires the command-line tool ffmpeg
to be installed on your system, which is available from most package managers:
# on Ubuntu or Debian
sudo apt update && sudo apt install ffmpeg
# on Arch Linux
sudo pacman -S ffmpeg
# on MacOS using Homebrew (https://brew.sh/)
brew install ffmpeg
# on Windows using Chocolatey (https://chocolatey.org/)
choco install ffmpeg
# on Windows using Scoop (https://scoop.sh/)
scoop install ffmpeg
You may need rust
installed as well, in case tiktoken does not provide a pre-built wheel for your platform. If you see installation errors during the pip install
command above, please follow the Getting started page to install Rust development environment. Additionally, you may need to configure the PATH
environment variable, e.g. export PATH="$HOME/.cargo/bin:$PATH"
. If the installation fails with No module named 'setuptools_rust'
, you need to install setuptools_rust
, e.g. by running:
pip install setuptools-rust
There are five model sizes, four with English-only versions, offering speed and accuracy tradeoffs. Below are the names of the available models and their approximate memory requirements and inference speed relative to the large model; actual speed may vary depending on many factors including the available hardware.
Size | Parameters | English-only model | Multilingual model | Required VRAM | Relative speed |
---|---|---|---|---|---|
tiny | 39 M | tiny.en |
tiny |
~1 GB | ~32x |
base | 74 M | base.en |
base |
~1 GB | ~16x |
small | 244 M | small.en |
small |
~2 GB | ~6x |
medium | 769 M | medium.en |
medium |
~5 GB | ~2x |
large | 1550 M | N/A | large |
~10 GB | 1x |
The .en
models for English-only applications tend to perform better, especially for the tiny.en
and base.en
models. We observed that the difference becomes less significant for the small.en
and medium.en
models.
Whisper's performance varies widely depending on the language. The figure below shows a performance breakdown of large-v3
and large-v2
models by language, using WERs (word error rates) or CER (character error rates, shown in Italic) evaluated on the Common Voice 15 and Fleurs datasets. Additional WER/CER metrics corresponding to the other models and datasets can be found in Appendix D.1, D.2, and D.4 of the paper, as well as the BLEU (Bilingual Evaluation Understudy) scores for translation in Appendix D.3.
The following command will transcribe speech in audio files, using the medium
model:
whisper audio.flac audio.mp3 audio.wav --model medium
The default setting (which selects the small
model) works well for transcribing English. To transcribe an audio file containing non-English speech, you can specify the language using the --language
option:
whisper japanese.wav --language Japanese
Adding --task translate
will translate the speech into English:
whisper japanese.wav --language Japanese --task translate
Run the following to view all available options:
whisper --help
See tokenizer.py for the list of all available languages.
Transcription can also be performed within Python:
import whisper
model = whisper.load_model("base")
result = model.transcribe("audio.mp3")
print(result["text"])
Internally, the transcribe()
method reads the entire file and processes the audio with a sliding 30-second window, performing autoregressive sequence-to-sequence predictions on each window.
Below is an example usage of whisper.detect_language()
and whisper.decode()
which provide lower-level access to the model.
import whisper
model = whisper.load_model("base")
# load audio and pad/trim it to fit 30 seconds
audio = whisper.load_audio("audio.mp3")
audio = whisper.pad_or_trim(audio)
# make log-Mel spectrogram and move to the same device as the model
mel = whisper.log_mel_spectrogram(audio).to(model.device)
# detect the spoken language
_, probs = model.detect_language(mel)
print(f"Detected language: {max(probs, key=probs.get)}")
# decode the audio
options = whisper.DecodingOptions()
result = whisper.decode(model, mel, options)
# print the recognized text
print(result.text)
Please use the 🙌 Show and tell category in Discussions for sharing more example usages of Whisper and third-party extensions such as web demos, integrations with other tools, ports for different platforms, etc.
Whisper's code and model weights are released under the MIT License. See LICENSE for further details.