base on Faster Whisper transcription with CTranslate2 [](https://github.com/SYSTRAN/faster-whisper/actions?query=workflow%3ACI) [](https://badge.fury.io/py/faster-whisper) # Faster Whisper transcription with CTranslate2 **faster-whisper** is a reimplementation of OpenAI's Whisper model using [CTranslate2](https://github.com/OpenNMT/CTranslate2/), which is a fast inference engine for Transformer models. This implementation is up to 4 times faster than [openai/whisper](https://github.com/openai/whisper) for the same accuracy while using less memory. The efficiency can be further improved with 8-bit quantization on both CPU and GPU. ## Benchmark ### Whisper For reference, here's the time and memory usage that are required to transcribe [**13 minutes**](https://www.youtube.com/watch?v=0u7tTptBo9I) of audio using different implementations: * [openai/whisper](https://github.com/openai/whisper)@[v20240930](https://github.com/openai/whisper/tree/v20240930) * [whisper.cpp](https://github.com/ggerganov/whisper.cpp)@[v1.7.2](https://github.com/ggerganov/whisper.cpp/tree/v1.7.2) * [transformers](https://github.com/huggingface/transformers)@[v4.46.3](https://github.com/huggingface/transformers/tree/v4.46.3) * [faster-whisper](https://github.com/SYSTRAN/faster-whisper)@[v1.1.0](https://github.com/SYSTRAN/faster-whisper/tree/v1.1.0) ### Large-v2 model on GPU | Implementation | Precision | Beam size | Time | VRAM Usage | | --- | --- | --- | --- | --- | | openai/whisper | fp16 | 5 | 2m23s | 4708MB | | whisper.cpp (Flash Attention) | fp16 | 5 | 1m05s | 4127MB | | transformers (SDPA)[^1] | fp16 | 5 | 1m52s | 4960MB | | faster-whisper | fp16 | 5 | 1m03s | 4525MB | | faster-whisper (`batch_size=8`) | fp16 | 5 | 17s | 6090MB | | faster-whisper | int8 | 5 | 59s | 2926MB | | faster-whisper (`batch_size=8`) | int8 | 5 | 16s | 4500MB | ### distil-whisper-large-v3 model on GPU | Implementation | Precision | Beam size | Time | YT Commons WER | | --- | --- | --- | --- | --- | | transformers (SDPA) (`batch_size=16`) | fp16 | 5 | 46m12s | 14.801 | | faster-whisper (`batch_size=16`) | fp16 | 5 | 25m50s | 13.527 | *GPU Benchmarks are Executed with CUDA 12.4 on a NVIDIA RTX 3070 Ti 8GB.* [^1]: transformers OOM for any batch size > 1 ### Small model on CPU | Implementation | Precision | Beam size | Time | RAM Usage | | --- | --- | --- | --- | --- | | openai/whisper | fp32 | 5 | 6m58s | 2335MB | | whisper.cpp | fp32 | 5 | 2m05s | 1049MB | | whisper.cpp (OpenVINO) | fp32 | 5 | 1m45s | 1642MB | | faster-whisper | fp32 | 5 | 2m37s | 2257MB | | faster-whisper (`batch_size=8`) | fp32 | 5 | 1m06s | 4230MB | | faster-whisper | int8 | 5 | 1m42s | 1477MB | | faster-whisper (`batch_size=8`) | int8 | 5 | 51s | 3608MB | *Executed with 8 threads on an Intel Core i7-12700K.* ## Requirements * Python 3.9 or greater Unlike openai-whisper, FFmpeg does **not** need to be installed on the system. The audio is decoded with the Python library [PyAV](https://github.com/PyAV-Org/PyAV) which bundles the FFmpeg libraries in its package. ### GPU GPU execution requires the following NVIDIA libraries to be installed: * [cuBLAS for CUDA 12](https://developer.nvidia.com/cublas) * [cuDNN 9 for CUDA 12](https://developer.nvidia.com/cudnn) **Note**: The latest versions of `ctranslate2` only support CUDA 12 and cuDNN 9. For CUDA 11 and cuDNN 8, the current workaround is downgrading to the `3.24.0` version of `ctranslate2`, for CUDA 12 and cuDNN 8, downgrade to the `4.4.0` version of `ctranslate2`, (This can be done with `pip install --force-reinstall ctranslate2==4.4.0` or specifying the version in a `requirements.txt`). There are multiple ways to install the NVIDIA libraries mentioned above. The recommended way is described in the official NVIDIA documentation, but we also suggest other installation methods below. <details> <summary>Other installation methods (click to expand)</summary> **Note:** For all these methods below, keep in mind the above note regarding CUDA versions. Depending on your setup, you may need to install the _CUDA 11_ versions of libraries that correspond to the CUDA 12 libraries listed in the instructions below. #### Use Docker The libraries (cuBLAS, cuDNN) are installed in this official NVIDIA CUDA Docker images: `nvidia/cuda:12.3.2-cudnn9-runtime-ubuntu22.04`. #### Install with `pip` (Linux only) On Linux these libraries can be installed with `pip`. Note that `LD_LIBRARY_PATH` must be set before launching Python. ```bash pip install nvidia-cublas-cu12 nvidia-cudnn-cu12==9.* export LD_LIBRARY_PATH=`python3 -c 'import os; import nvidia.cublas.lib; import nvidia.cudnn.lib; print(os.path.dirname(nvidia.cublas.lib.__file__) + ":" + os.path.dirname(nvidia.cudnn.lib.__file__))'` ``` #### Download the libraries from Purfview's repository (Windows & Linux) Purfview's [whisper-standalone-win](https://github.com/Purfview/whisper-standalone-win) provides the required NVIDIA libraries for Windows & Linux in a [single archive](https://github.com/Purfview/whisper-standalone-win/releases/tag/libs). Decompress the archive and place the libraries in a directory included in the `PATH`. </details> ## Installation The module can be installed from [PyPI](https://pypi.org/project/faster-whisper/): ```bash pip install faster-whisper ``` <details> <summary>Other installation methods (click to expand)</summary> ### Install the master branch ```bash pip install --force-reinstall "faster-whisper @ https://github.com/SYSTRAN/faster-whisper/archive/refs/heads/master.tar.gz" ``` ### Install a specific commit ```bash pip install --force-reinstall "faster-whisper @ https://github.com/SYSTRAN/faster-whisper/archive/a4f1cc8f11433e454c3934442b5e1a4ed5e865c3.tar.gz" ``` </details> ## Usage ### Faster-whisper ```python from faster_whisper import WhisperModel model_size = "large-v3" # Run on GPU with FP16 model = WhisperModel(model_size, device="cuda", compute_type="float16") # or run on GPU with INT8 # model = WhisperModel(model_size, device="cuda", compute_type="int8_float16") # or run on CPU with INT8 # model = WhisperModel(model_size, device="cpu", compute_type="int8") segments, info = model.transcribe("audio.mp3", beam_size=5) print("Detected language '%s' with probability %f" % (info.language, info.language_probability)) for segment in segments: print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text)) ``` **Warning:** `segments` is a *generator* so the transcription only starts when you iterate over it. The transcription can be run to completion by gathering the segments in a list or a `for` loop: ```python segments, _ = model.transcribe("audio.mp3") segments = list(segments) # The transcription will actually run here. ``` ### Batched Transcription The following code snippet illustrates how to run batched transcription on an example audio file. `BatchedInferencePipeline.transcribe` is a drop-in replacement for `WhisperModel.transcribe` ```python from faster_whisper import WhisperModel, BatchedInferencePipeline model = WhisperModel("turbo", device="cuda", compute_type="float16") batched_model = BatchedInferencePipeline(model=model) segments, info = batched_model.transcribe("audio.mp3", batch_size=16) for segment in segments: print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text)) ``` ### Faster Distil-Whisper The Distil-Whisper checkpoints are compatible with the Faster-Whisper package. In particular, the latest [distil-large-v3](https://huggingface.co/distil-whisper/distil-large-v3) checkpoint is intrinsically designed to work with the Faster-Whisper transcription algorithm. The following code snippet demonstrates how to run inference with distil-large-v3 on a specified audio file: ```python from faster_whisper import WhisperModel model_size = "distil-large-v3" model = WhisperModel(model_size, device="cuda", compute_type="float16") segments, info = model.transcribe("audio.mp3", beam_size=5, language="en", condition_on_previous_text=False) for segment in segments: print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text)) ``` For more information about the distil-large-v3 model, refer to the original [model card](https://huggingface.co/distil-whisper/distil-large-v3). ### Word-level timestamps ```python segments, _ = model.transcribe("audio.mp3", word_timestamps=True) for segment in segments: for word in segment.words: print("[%.2fs -> %.2fs] %s" % (word.start, word.end, word.word)) ``` ### VAD filter The library integrates the [Silero VAD](https://github.com/snakers4/silero-vad) model to filter out parts of the audio without speech: ```python segments, _ = model.transcribe("audio.mp3", vad_filter=True) ``` The default behavior is conservative and only removes silence longer than 2 seconds. See the available VAD parameters and default values in the [source code](https://github.com/SYSTRAN/faster-whisper/blob/master/faster_whisper/vad.py). They can be customized with the dictionary argument `vad_parameters`: ```python segments, _ = model.transcribe( "audio.mp3", vad_filter=True, vad_parameters=dict(min_silence_duration_ms=500), ) ``` Vad filter is enabled by default for batched transcription. ### Logging The library logging level can be configured like this: ```python import logging logging.basicConfig() logging.getLogger("faster_whisper").setLevel(logging.DEBUG) ``` ### Going further See more model and transcription options in the [`WhisperModel`](https://github.com/SYSTRAN/faster-whisper/blob/master/faster_whisper/transcribe.py) class implementation. ## Community integrations Here is a non exhaustive list of open-source projects using faster-whisper. Feel free to add your project to the list! * [faster-whisper-server](https://github.com/fedirz/faster-whisper-server) is an OpenAI compatible server using `faster-whisper`. It's easily deployable with Docker, works with OpenAI SDKs/CLI, supports streaming, and live transcription. * [WhisperX](https://github.com/m-bain/whisperX) is an award-winning Python library that offers speaker diarization and accurate word-level timestamps using wav2vec2 alignment * [whisper-ctranslate2](https://github.com/Softcatala/whisper-ctranslate2) is a command line client based on faster-whisper and compatible with the original client from openai/whisper. * [whisper-diarize](https://github.com/MahmoudAshraf97/whisper-diarization) is a speaker diarization tool that is based on faster-whisper and NVIDIA NeMo. * [whisper-standalone-win](https://github.com/Purfview/whisper-standalone-win) Standalone CLI executables of faster-whisper for Windows, Linux & macOS. * [asr-sd-pipeline](https://github.com/hedrergudene/asr-sd-pipeline) provides a scalable, modular, end to end multi-speaker speech to text solution implemented using AzureML pipelines. * [Open-Lyrics](https://github.com/zh-plus/Open-Lyrics) is a Python library that transcribes voice files using faster-whisper, and translates/polishes the resulting text into `.lrc` files in the desired language using OpenAI-GPT. * [wscribe](https://github.com/geekodour/wscribe) is a flexible transcript generation tool supporting faster-whisper, it can export word level transcript and the exported transcript then can be edited with [wscribe-editor](https://github.com/geekodour/wscribe-editor) * [aTrain](https://github.com/BANDAS-Center/aTrain) is a graphical user interface implementation of faster-whisper developed at the BANDAS-Center at the University of Graz for transcription and diarization in Windows ([Windows Store App](https://apps.microsoft.com/detail/atrain/9N15Q44SZNS2)) and Linux. * [Whisper-Streaming](https://github.com/ufal/whisper_streaming) implements real-time mode for offline Whisper-like speech-to-text models with faster-whisper as the most recommended back-end. It implements a streaming policy with self-adaptive latency based on the actual source complexity, and demonstrates the state of the art. * [WhisperLive](https://github.com/collabora/WhisperLive) is a nearly-live implementation of OpenAI's Whisper which uses faster-whisper as the backend to transcribe audio in real-time. * [Faster-Whisper-Transcriber](https://github.com/BBC-Esq/ctranslate2-faster-whisper-transcriber) is a simple but reliable voice transcriber that provides a user-friendly interface. * [Open-dubbing](https://github.com/softcatala/open-dubbing) is open dubbing is an AI dubbing system which uses machine learning models to automatically translate and synchronize audio dialogue into different languages. ## Model conversion When loading a model from its size such as `WhisperModel("large-v3")`, the corresponding CTranslate2 model is automatically downloaded from the [Hugging Face Hub](https://huggingface.co/Systran). We also provide a script to convert any Whisper models compatible with the Transformers library. They could be the original OpenAI models or user fine-tuned models. For example the command below converts the [original "large-v3" Whisper model](https://huggingface.co/openai/whisper-large-v3) and saves the weights in FP16: ```bash pip install transformers[torch]>=4.23 ct2-transformers-converter --model openai/whisper-large-v3 --output_dir whisper-large-v3-ct2 --copy_files tokenizer.json preprocessor_config.json --quantization float16 ``` * The option `--model` accepts a model name on the Hub or a path to a model directory. * If the option `--copy_files tokenizer.json` is not used, the tokenizer configuration is automatically downloaded when the model is loaded later. Models can also be converted from the code. See the [conversion API](https://opennmt.net/CTranslate2/python/ctranslate2.converters.TransformersConverter.html). ### Load a converted model 1. Directly load the model from a local directory: ```python model = faster_whisper.WhisperModel("whisper-large-v3-ct2") ``` 2. [Upload your model to the Hugging Face Hub](https://huggingface.co/docs/transformers/model_sharing#upload-with-the-web-interface) and load it from its name: ```python model = faster_whisper.WhisperModel("username/whisper-large-v3-ct2") ``` ## Comparing performance against other implementations If you are comparing the performance against other Whisper implementations, you should make sure to run the comparison with similar settings. In particular: * Verify that the same transcription options are used, especially the same beam size. For example in openai/whisper, `model.transcribe` uses a default beam size of 1 but here we use a default beam size of 5. * Transcription speed is closely affected by the number of words in the transcript, so ensure that other implementations have a similar WER (Word Error Rate) to this one. * When running on CPU, make sure to set the same number of threads. Many frameworks will read the environment variable `OMP_NUM_THREADS`, which can be set when running your script: ```bash OMP_NUM_THREADS=4 python3 my_script.py ``` ", Assign "at most 3 tags" to the expected json: {"id":"4998","tags":[]} "only from the tags list I provide: [{"id":77,"name":"3d"},{"id":89,"name":"agent"},{"id":17,"name":"ai"},{"id":54,"name":"algorithm"},{"id":24,"name":"api"},{"id":44,"name":"authentication"},{"id":3,"name":"aws"},{"id":27,"name":"backend"},{"id":60,"name":"benchmark"},{"id":72,"name":"best-practices"},{"id":39,"name":"bitcoin"},{"id":37,"name":"blockchain"},{"id":1,"name":"blog"},{"id":45,"name":"bundler"},{"id":58,"name":"cache"},{"id":21,"name":"chat"},{"id":49,"name":"cicd"},{"id":4,"name":"cli"},{"id":64,"name":"cloud-native"},{"id":48,"name":"cms"},{"id":61,"name":"compiler"},{"id":68,"name":"containerization"},{"id":92,"name":"crm"},{"id":34,"name":"data"},{"id":47,"name":"database"},{"id":8,"name":"declarative-gui "},{"id":9,"name":"deploy-tool"},{"id":53,"name":"desktop-app"},{"id":6,"name":"dev-exp-lib"},{"id":59,"name":"dev-tool"},{"id":13,"name":"ecommerce"},{"id":26,"name":"editor"},{"id":66,"name":"emulator"},{"id":62,"name":"filesystem"},{"id":80,"name":"finance"},{"id":15,"name":"firmware"},{"id":73,"name":"for-fun"},{"id":2,"name":"framework"},{"id":11,"name":"frontend"},{"id":22,"name":"game"},{"id":81,"name":"game-engine "},{"id":23,"name":"graphql"},{"id":84,"name":"gui"},{"id":91,"name":"http"},{"id":5,"name":"http-client"},{"id":51,"name":"iac"},{"id":30,"name":"ide"},{"id":78,"name":"iot"},{"id":40,"name":"json"},{"id":83,"name":"julian"},{"id":38,"name":"k8s"},{"id":31,"name":"language"},{"id":10,"name":"learning-resource"},{"id":33,"name":"lib"},{"id":41,"name":"linter"},{"id":28,"name":"lms"},{"id":16,"name":"logging"},{"id":76,"name":"low-code"},{"id":90,"name":"message-queue"},{"id":42,"name":"mobile-app"},{"id":18,"name":"monitoring"},{"id":36,"name":"networking"},{"id":7,"name":"node-version"},{"id":55,"name":"nosql"},{"id":57,"name":"observability"},{"id":46,"name":"orm"},{"id":52,"name":"os"},{"id":14,"name":"parser"},{"id":74,"name":"react"},{"id":82,"name":"real-time"},{"id":56,"name":"robot"},{"id":65,"name":"runtime"},{"id":32,"name":"sdk"},{"id":71,"name":"search"},{"id":63,"name":"secrets"},{"id":25,"name":"security"},{"id":85,"name":"server"},{"id":86,"name":"serverless"},{"id":70,"name":"storage"},{"id":75,"name":"system-design"},{"id":79,"name":"terminal"},{"id":29,"name":"testing"},{"id":12,"name":"ui"},{"id":50,"name":"ux"},{"id":88,"name":"video"},{"id":20,"name":"web-app"},{"id":35,"name":"web-server"},{"id":43,"name":"webassembly"},{"id":69,"name":"workflow"},{"id":87,"name":"yaml"}]" returns me the "expected json"