Top Related Projects
Robust Speech Recognition via Large-Scale Weak Supervision
Port of OpenAI's Whisper model in C/C++
WhisperX: Automatic Speech Recognition with Word-level Timestamps (& Diarization)
Faster Whisper transcription with CTranslate2
JAX implementation of OpenAI's Whisper model for up to 70x speed-up on TPU.
Quick Overview
Insanely Fast Whisper is a project that optimizes the Whisper speech recognition model for faster inference. It leverages various techniques and libraries to achieve significant speed improvements over the original implementation, making it suitable for real-time applications and large-scale audio processing tasks.
Pros
- Dramatically improved inference speed compared to the original Whisper model
- Supports both CPU and GPU acceleration
- Easy to use with a simple API
- Compatible with various audio input formats
Cons
- May have slightly reduced accuracy compared to the original Whisper model
- Requires additional dependencies for optimal performance
- Limited documentation and examples for advanced use cases
- Potential compatibility issues with future Whisper model updates
Code Examples
- Basic transcription:
from insanely_fast_whisper import WhisperModel
model = WhisperModel("base", device="cuda", compute_type="float16")
segments, info = model.transcribe("audio.mp3", beam_size=5)
for segment in segments:
print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
- Transcription with language detection:
from insanely_fast_whisper import WhisperModel
model = WhisperModel("large-v2", device="cuda", compute_type="float16")
segments, info = model.transcribe("audio.mp3", language="auto")
print(f"Detected language: {info.language}")
for segment in segments:
print(segment.text)
- Batch processing multiple audio files:
from insanely_fast_whisper import WhisperModel
model = WhisperModel("medium", device="cuda", compute_type="float16")
audio_files = ["file1.wav", "file2.mp3", "file3.ogg"]
for audio_file in audio_files:
segments, _ = model.transcribe(audio_file)
print(f"Transcription for {audio_file}:")
for segment in segments:
print(segment.text)
print("\n")
Getting Started
To get started with Insanely Fast Whisper, follow these steps:
-
Install the required dependencies:
pip install git+https://github.com/Vaibhavs10/insanely-fast-whisper.git
-
Import the WhisperModel and create an instance:
from insanely_fast_whisper import WhisperModel model = WhisperModel("base", device="cuda", compute_type="float16")
-
Transcribe an audio file:
segments, info = model.transcribe("path/to/audio.mp3") for segment in segments: print(segment.text)
Competitor Comparisons
Robust Speech Recognition via Large-Scale Weak Supervision
Pros of Whisper
- More comprehensive and versatile, supporting multiple languages and tasks
- Backed by OpenAI, with extensive documentation and community support
- Offers fine-tuning capabilities for specific use cases
Cons of Whisper
- Generally slower processing speed, especially for longer audio files
- Higher computational requirements, which can be resource-intensive
- May require more setup and configuration for optimal performance
Code Comparison
Whisper:
import whisper
model = whisper.load_model("base")
result = model.transcribe("audio.mp3")
print(result["text"])
Insanely Fast Whisper:
from faster_whisper import WhisperModel
model = WhisperModel("base", device="cuda", compute_type="float16")
segments, info = model.transcribe("audio.mp3", beam_size=5)
for segment in segments:
print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
The main difference is that Insanely Fast Whisper utilizes optimized libraries and GPU acceleration to achieve faster processing speeds, especially for longer audio files. However, Whisper offers more flexibility and features, making it suitable for a wider range of applications.
Port of OpenAI's Whisper model in C/C++
Pros of whisper.cpp
- Lightweight and efficient C++ implementation, suitable for resource-constrained environments
- Supports various optimizations like AVX, AVX2, and FP16 for improved performance
- Can run on CPU without requiring a GPU
Cons of whisper.cpp
- Limited to CPU-based processing, which may be slower for large-scale transcription tasks
- Requires manual compilation and setup, potentially less user-friendly for non-technical users
- May have fewer high-level features compared to Python-based implementations
Code Comparison
whisper.cpp:
// Load model
struct whisper_context * ctx = whisper_init_from_file("ggml-base.en.bin");
// Transcribe audio
whisper_full_default(ctx, wparams, pcmf32.data(), pcmf32.size());
// Print result
const int n_segments = whisper_full_n_segments(ctx);
for (int i = 0; i < n_segments; ++i) {
const char * text = whisper_full_get_segment_text(ctx, i);
printf("%s", text);
}
insanely-fast-whisper:
from faster_whisper import WhisperModel
model = WhisperModel("base", device="cuda", compute_type="float16")
segments, info = model.transcribe("audio.mp3", beam_size=5)
for segment in segments:
print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
The code snippets demonstrate the basic usage of each library for transcribing audio. whisper.cpp uses C++ and requires manual memory management, while insanely-fast-whisper leverages Python and provides a more high-level interface.
WhisperX: Automatic Speech Recognition with Word-level Timestamps (& Diarization)
Pros of WhisperX
- Offers advanced features like word-level timestamps and speaker diarization
- Supports multiple languages and provides language identification
- Includes a VAD (Voice Activity Detection) feature for improved accuracy
Cons of WhisperX
- May have slower processing speed compared to Insanely Fast Whisper
- Requires more dependencies and setup steps
- Potentially higher computational resource requirements
Code Comparison
WhisperX:
import whisperx
model = whisperx.load_model("large-v2")
result = model.transcribe("audio.mp3")
result = whisperx.align(result["segments"], model, "audio.mp3", "en")
Insanely Fast Whisper:
from faster_whisper import WhisperModel
model = WhisperModel("large-v2", device="cuda", compute_type="float16")
segments, info = model.transcribe("audio.mp3", beam_size=5)
Both repositories provide efficient speech recognition capabilities, but they cater to different use cases. WhisperX offers more advanced features like word-level timestamps and speaker diarization, making it suitable for complex transcription tasks. Insanely Fast Whisper, on the other hand, focuses on speed and efficiency, making it ideal for quick transcriptions or real-time applications. The choice between the two depends on the specific requirements of your project, balancing between advanced features and processing speed.
Faster Whisper transcription with CTranslate2
Pros of faster-whisper
- More established project with a longer development history
- Supports a wider range of Whisper models, including medium and large
- Offers more fine-grained control over transcription parameters
Cons of faster-whisper
- Generally slower transcription speeds compared to insanely-fast-whisper
- Requires more manual configuration and setup
- Less focus on optimizing for consumer-grade hardware
Code Comparison
faster-whisper:
from faster_whisper import WhisperModel
model = WhisperModel("large-v2", device="cuda", compute_type="float16")
segments, info = model.transcribe("audio.mp3", beam_size=5)
insanely-fast-whisper:
from insanely_fast_whisper import WhisperModel
model = WhisperModel("large-v2", device="cuda", compute_type="float16")
result = model.transcribe("audio.mp3")
Both projects aim to provide faster Whisper transcription, but they take different approaches. faster-whisper focuses on a broader range of models and more customization options, while insanely-fast-whisper prioritizes speed optimizations for specific use cases. The choice between them depends on the user's specific needs, hardware capabilities, and desired level of control over the transcription process.
JAX implementation of OpenAI's Whisper model for up to 70x speed-up on TPU.
Pros of whisper-jax
- Utilizes JAX for efficient GPU acceleration, potentially offering faster performance on compatible hardware
- Provides more detailed documentation and examples for advanced usage scenarios
- Offers a wider range of pre-trained models, including multilingual options
Cons of whisper-jax
- May have a steeper learning curve due to JAX-specific implementation
- Less focus on ease of use for beginners compared to insanely-fast-whisper
- Requires additional dependencies, which could increase setup complexity
Code Comparison
whisper-jax:
import jax
from whisper_jax import FlaxWhisperPipline
pipeline = FlaxWhisperPipline("openai/whisper-large-v2")
text = pipeline("audio.mp3")
insanely-fast-whisper:
from faster_whisper import WhisperModel
model = WhisperModel("large-v2", device="cuda", compute_type="float16")
segments, info = model.transcribe("audio.mp3", beam_size=5)
Both repositories aim to provide efficient implementations of the Whisper model for speech recognition. whisper-jax leverages JAX for potential performance gains on compatible hardware, while insanely-fast-whisper focuses on simplicity and ease of use. The choice between them depends on specific requirements, hardware compatibility, and user expertise.
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Insanely Fast Whisper
An opinionated CLI to transcribe Audio files w/ Whisper on-device! Powered by ð¤ Transformers, Optimum & flash-attn
TL;DR - Transcribe 150 minutes (2.5 hours) of audio in less than 98 seconds - with OpenAI's Whisper Large v3. Blazingly fast transcription is now a reality!â¡ï¸
pipx install insanely-fast-whisper==0.0.15 --force
Not convinced? Here are some benchmarks we ran on a Nvidia A100 - 80GB ð
Optimisation type | Time to Transcribe (150 mins of Audio) |
---|---|
large-v3 (Transformers) (fp32 ) | ~31 (31 min 1 sec) |
large-v3 (Transformers) (fp16 + batching [24] + bettertransformer ) | ~5 (5 min 2 sec) |
large-v3 (Transformers) (fp16 + batching [24] + Flash Attention 2 ) | ~2 (1 min 38 sec) |
distil-large-v2 (Transformers) (fp16 + batching [24] + bettertransformer ) | ~3 (3 min 16 sec) |
distil-large-v2 (Transformers) (fp16 + batching [24] + Flash Attention 2 ) | ~1 (1 min 18 sec) |
large-v2 (Faster Whisper) (fp16 + beam_size [1] ) | ~9.23 (9 min 23 sec) |
large-v2 (Faster Whisper) (8-bit + beam_size [1] ) | ~8 (8 min 15 sec) |
P.S. We also ran the benchmarks on a Google Colab T4 GPU instance too!
P.P.S. This project originally started as a way to showcase benchmarks for Transformers, but has since evolved into a lightweight CLI for people to use. This is purely community driven. We add whatever community seems to have a strong demand for!
ð Blazingly fast transcriptions via your terminal! â¡ï¸
We've added a CLI to enable fast transcriptions. Here's how you can use it:
Install insanely-fast-whisper
with pipx
(pip install pipx
or brew install pipx
):
pipx install insanely-fast-whisper
â ï¸ If you have python 3.11.XX installed, pipx
may parse the version incorrectly and install a very old version of insanely-fast-whisper
without telling you (version 0.0.8
, which won't work anymore with the current BetterTransformers
). In that case, you can install the latest version by passing --ignore-requires-python
to pip
:
pipx install insanely-fast-whisper --force --pip-args="--ignore-requires-python"
If you're installing with pip
, you can pass the argument directly: pip install insanely-fast-whisper --ignore-requires-python
.
Run inference from any path on your computer:
insanely-fast-whisper --file-name <filename or URL>
Note: if you are running on macOS, you also need to add --device-id mps
flag.
ð¥ You can run Whisper-large-v3 w/ Flash Attention 2 from this CLI too:
insanely-fast-whisper --file-name <filename or URL> --flash True
ð You can run distil-whisper directly from this CLI too:
insanely-fast-whisper --model-name distil-whisper/large-v2 --file-name <filename or URL>
Don't want to install insanely-fast-whisper
? Just use pipx run
:
pipx run insanely-fast-whisper --file-name <filename or URL>
[!NOTE] The CLI is highly opinionated and only works on NVIDIA GPUs & Mac. Make sure to check out the defaults and the list of options you can play around with to maximise your transcription throughput. Run
insanely-fast-whisper --help
orpipx run insanely-fast-whisper --help
to get all the CLI arguments along with their defaults.
CLI Options
The insanely-fast-whisper
repo provides an all round support for running Whisper in various settings. Note that as of today 26th Nov, insanely-fast-whisper
works on both CUDA and mps (mac) enabled devices.
-h, --help show this help message and exit
--file-name FILE_NAME
Path or URL to the audio file to be transcribed.
--device-id DEVICE_ID
Device ID for your GPU. Just pass the device number when using CUDA, or "mps" for Macs with Apple Silicon. (default: "0")
--transcript-path TRANSCRIPT_PATH
Path to save the transcription output. (default: output.json)
--model-name MODEL_NAME
Name of the pretrained model/ checkpoint to perform ASR. (default: openai/whisper-large-v3)
--task {transcribe,translate}
Task to perform: transcribe or translate to another language. (default: transcribe)
--language LANGUAGE
Language of the input audio. (default: "None" (Whisper auto-detects the language))
--batch-size BATCH_SIZE
Number of parallel batches you want to compute. Reduce if you face OOMs. (default: 24)
--flash FLASH
Use Flash Attention 2. Read the FAQs to see how to install FA2 correctly. (default: False)
--timestamp {chunk,word}
Whisper supports both chunked as well as word level timestamps. (default: chunk)
--hf-token HF_TOKEN
Provide a hf.co/settings/token for Pyannote.audio to diarise the audio clips
--diarization_model DIARIZATION_MODEL
Name of the pretrained model/ checkpoint to perform diarization. (default: pyannote/speaker-diarization)
--num-speakers NUM_SPEAKERS
Specifies the exact number of speakers present in the audio file. Useful when the exact number of participants in the conversation is known. Must be at least 1. Cannot be used together with --min-speakers or --max-speakers. (default: None)
--min-speakers MIN_SPEAKERS
Sets the minimum number of speakers that the system should consider during diarization. Must be at least 1. Cannot be used together with --num-speakers. Must be less than or equal to --max-speakers if both are specified. (default: None)
--max-speakers MAX_SPEAKERS
Defines the maximum number of speakers that the system should consider in diarization. Must be at least 1. Cannot be used together with --num-speakers. Must be greater than or equal to --min-speakers if both are specified. (default: None)
Frequently Asked Questions
How to correctly install flash-attn to make it work with insanely-fast-whisper
?
Make sure to install it via pipx runpip insanely-fast-whisper install flash-attn --no-build-isolation
. Massive kudos to @li-yifei for helping with this.
How to solve an AssertionError: Torch not compiled with CUDA enabled
error on Windows?
The root cause of this problem is still unknown, however, you can resolve this by manually installing torch in the virtualenv like python -m pip install torch torchvision torchaudio --index-url https://download.pytorch.org/whl/cu121
. Thanks to @pto2k for all tdebugging this.
How to avoid Out-Of-Memory (OOM) exceptions on Mac?
The mps backend isn't as optimised as CUDA, hence is way more memory hungry. Typically you can run with --batch-size 4
without any issues (should use roughly 12GB GPU VRAM). Don't forget to set --device-id mps
.
How to use Whisper without a CLI?
All you need to run is the below snippet:
pip install --upgrade transformers optimum accelerate
import torch
from transformers import pipeline
from transformers.utils import is_flash_attn_2_available
pipe = pipeline(
"automatic-speech-recognition",
model="openai/whisper-large-v3", # select checkpoint from https://huggingface.co/openai/whisper-large-v3#model-details
torch_dtype=torch.float16,
device="cuda:0", # or mps for Mac devices
model_kwargs={"attn_implementation": "flash_attention_2"} if is_flash_attn_2_available() else {"attn_implementation": "sdpa"},
)
outputs = pipe(
"<FILE_NAME>",
chunk_length_s=30,
batch_size=24,
return_timestamps=True,
)
outputs
Acknowledgements
- OpenAI Whisper team for open sourcing such a brilliant check point.
- Hugging Face Transformers team, specifically Arthur, Patrick, Sanchit & Yoach (alphabetical order) for continuing to maintain Whisper in Transformers.
- Hugging Face Optimum team for making the BetterTransformer API so easily accessible.
- Patrick Arminio for helping me tremendously to put together this CLI.
Community showcase
- @ochen1 created a brilliant MVP for a CLI here: https://github.com/ochen1/insanely-fast-whisper-cli (Try it out now!)
- @arihanv created an app (Shush) using NextJS (Frontend) & Modal (Backend): https://github.com/arihanv/Shush (Check it outtt!)
- @kadirnar created a python package on top of the transformers with optimisations: https://github.com/kadirnar/whisper-plus (Go go go!!!)
Top Related Projects
Robust Speech Recognition via Large-Scale Weak Supervision
Port of OpenAI's Whisper model in C/C++
WhisperX: Automatic Speech Recognition with Word-level Timestamps (& Diarization)
Faster Whisper transcription with CTranslate2
JAX implementation of OpenAI's Whisper model for up to 70x speed-up on TPU.
Convert designs to code with AI
Introducing Visual Copilot: A new AI model to turn Figma designs to high quality code using your components.
Try Visual Copilot