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Faster Whisper
by SYSTRANPython
Reimplementation of Whisper up to 4x faster with less memory.
Maturity: slowing because last commit 9 months ago. Derived from release and commit history, not a rating.
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- Last commit
- 2025-11-19
- License
- MIT
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In practice
Written by AI from this repository’s README · high confidenceReference Whisper inference is slow and memory hungry, with no batching and no quantization option.
Use it when
When you need Whisper accuracy with substantially faster inference and optional 8-bit quantization on CPU or GPU.
Not the right pick when
GPU execution requires cuBLAS for CUDA 12 and cuDNN 9, and older CUDA setups need a downgraded ctranslate2.
Capabilities
- up to 4 times faster than openai/whisper for the same accuracy
- 8-bit quantization on both CPU and GPU
- batched transcription controlled by batch_size
- audio decoded with PyAV so system FFmpeg is not required
- published time and memory benchmarks for GPU and CPU
Requirements
- Python 3.9 or greater
- cuBLAS for CUDA 12 and cuDNN 9 for GPU execution
Cost: Free and open source
Install
Derived from the published package name in the repository, not from a model.
Video walkthroughs
Open Source Faster Whisper Voice transcription model running locally. Install instructions included
Convert speech to text in realtime without delay | using faster-whisper module
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What the repository ships
Detected from the actual files in the repository root.
Latest release v1.2.1
Published 2025-10-31
What's Changed
- only merge when
clip_timestampsare not provided by @MahmoudAshraf97 in https://github.com/SYSTRAN/faster-whisper/pull/1345 - Fix: Prevent <|nocaptions|> tokens in BatchedInferencePipeline by @mmichelli in https://github.com/SYSTRAN/faster-whisper/pull/1338
- Upgrade to Silero-VAD V6 by @MahmoudAshraf97 and @sssshhhhhh in https://github.com/SYSTRAN/faster-whisper/pull/1373
- Offload retry logic to hf hub by @MahmoudAshraf97 in https://github.com/SYSTRAN/faster-whisper/pull/1382
- Refinements for clip timestamps in batched inference by @MahmoudAshraf97 in https://github.com/SYSTRAN/faster-whisper/pull/1376
New Contributors
- @mmichelli made their first contribution in https://github.com/SYSTRAN/faster-whisper/pull/1338
- @sssshhhhhh made their first contribution in https://github.com/SYSTRAN/faster-whisper/pull/1365
Full Changelog: https://github.com/SYSTRAN/faster-whisper/compare/v1.2.0...v1.2.1
Tags
README
Faster Whisper transcription with CTranslate2
faster-whisper is a reimplementation of OpenAI's Whisper model using CTranslate2, which is a fast inference engine for Transformer models.
This implementation is up to 4 times faster than 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 of audio using different implementations:
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 which bundles the FFmpeg libraries in its package.
GPU
GPU execution requires the following NVIDIA libraries to be installed:
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.
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 provides the required NVIDIA libraries for Windows & Linux in a single archive. Decompress the archive and place the libraries in a directory included in the PATH.
</details>
Installation
The module can be installed from PyPI:
pip install faster-whisper
<details>
<summary>Other installation methods (click to expand)</summary>
Install the master branch
pip install --force-reinstall "faster-whisper @ https://github.com/SYSTRAN/faster-whisper/archive/refs/heads/master.tar.gz"
Install a specific commit
pip install --force-reinstall "faster-whisper @ https://github.com/SYSTRAN/faster-whisper/archive/a4f1cc8f11433e454c3934442b5e1a4ed5e865c3.tar.gz"
</details>
Usage
Faster-whisper
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:
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
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
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:
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.
Word-level timestamps
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 model to filter out parts of the audio without speech:
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. They can be customized with the dictionary argument vad_parameters:
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:
import logging
logging.basicConfig()
logging.getLogger("faster_whisper").setLevel(logging.DEBUG)
Going further
See more model and transcription options in the WhisperModel 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!
- speaches 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 is an award-winning Python library that offers speaker diarization and accurate word-level timestamps using wav2vec2 alignment
- whisper-ctranslate2 is a command line client based on faster-whisper and compatible with the original client from openai/whisper.
- whisper-diarize is a speaker diarization tool that is based on faster-whisper and NVIDIA NeMo.
- whisper-standalone-win Standalone CLI executables of faster-whisper for Windows, Linux & macOS.
- asr-sd-pipeline provides a scalable, modular, end to end multi-speaker speech to text solution implemented using AzureML pipelines.
- Open-Lyrics is a Python library that transcribes voice files using faster-whisper, and translates/polishes the resulting text into
.lrcfiles in the desired language using OpenAI-GPT. - 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
- 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) and Linux.
- 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 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 is a simple but reliable voice transcriber that provides a user-friendly interface.
- 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.
- Whisper-FastAPI whisper-fastapi is a very simple script that provides an API backend compatible with OpenAI, HomeAssistant, and Konele (Android voice typing) formats.
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.
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 and saves the weights in FP16:
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
--modelaccepts a model name on the Hub or a path to a model directory. - If the option
--copy_files tokenizer.jsonis 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.Transfor
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