Bonsai-27B-GGUF
lmstudio-community · Kit
A GGUF conversion of the 27B Bonsai model for LM Studio, with a 3.80 GB Q1_0 language file and a 0.93 GB BF16 vision projector. Tested for text, vision and par…
OpenAI's faster Whisper model for speech recognition and speech translation, with 809M parameters and only 4 decoding layers. For developers and anyone who transcribes audio.
whisper-large-v3-turbo is a kit in the Developer category, listed by Mindket. It is free. This is version 1, last updated on . Download it on this page, a zip of its files. Nothing in it runs by itself.
pip install --upgrade pip pip install --upgrade transformers datasets[audio] accelerate
whisper-large-v3-turbo is a speech recognition model from OpenAI. It transcribes spoken audio into text and can also translate speech into English. It has 809 million parameters.
It is a fine-tuned, pruned version of Whisper large-v3, with the decoding layers cut from 32 to 4. The card says this makes it much faster, with a small loss in quality.
Developers and teams who need to turn recordings into text, and anyone who wants Whisper quality at a faster speed on their own machine.
| Industry | Who uses it | How they use it | Example |
|---|---|---|---|
| Media | Journalists and editors | Transcribe interviews and recordings | Turn an audio file into text |
| Software | Developers | Add speech to text to an app | Run it through a Transformers pipeline |
| Education | Teachers and students | Caption lectures and talks | Get timestamps for each sentence |
Skip it if you need the best accuracy, since the card says the speed gain costs a small drop in quality.
| Use case | Who | What they get back |
|---|---|---|
| Transcription | Anyone with recordings | The spoken words as text |
| Translation to English | Multilingual teams | English text from speech in another language |
| Timestamps | Caption makers | Time marks for sentences or for single words |
pip install --upgrade pip pip install --upgrade transformers datasets[audio] accelerate
Create a speech recognition pipeline, then call it with the path of your audio file. A list of files with a batch size lets it transcribe several at once.
import torch
device = "cuda:0" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
model_id = "openai/whisper-large-v3-turbo"
model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
)
model.to(device)
processor = AutoProcessor.from_pretrained(model_id)
pipe = pipeline(
"automatic-speech-recognition",
model=model,
tokenizer=processor.tokenizer,
feature_extractor=processor.feature_extractor,
torch_dtype=torch_dtype,
device=device,
)
result = pipe("audio.mp3")
print(result["text"])Put your own value in its key setting first.
| Option | What it does |
|---|---|
| pipe("audio.mp3") | Transcribes one local file |
| pipe(list, batch_size=2) | Transcribes several files in parallel |
| language in generate_kwargs | Tells the model the language when you already know it |
| task set to translate | Gives English text from speech in another language |
| return_timestamps=True | Adds sentence-level timestamps |
| return_timestamps="word" | Adds word-level timestamps |
You pass the path of an interview recording to the pipeline and print the text. Add the word-level timestamp option and you also get a time mark for each word.
If you are new to speech recognition, run the file example first and check the text against what you hear. Then try timestamps.
If you know the field, the card says Transformers supports the Whisper decoding strategies, including temperature fallback and conditioning on previous tokens. Whisper predicts the language itself unless you pass one.
OpenAI's faster Whisper model for speech recognition and speech translation, with 809M parameters and only 4 decoding layers. For developers and anyone who transcribes audio.
Made by openai on Hugging Face. These notes are Mindket's, written from its model page and model card; the model's own files and license are on Hugging Face.
What it is: whisper-large-v3-turbo is a speech recognition model from OpenAI. It transcribes spoken audio into text and can also translate speech into English. It has 809 million parameters.
It is a fine-tuned, pruned version of Whisper large-v3, with the decoding layers cut from 32 to 4. The card says this makes it much faster, with a small loss in quality.
Who it suits: Developers and teams who need to turn recordings into text, and anyone who wants Whisper quality at a faster speed on their own machine.
Industry | Who uses it | How they use it | Example
Media | Journalists and editors | Transcribe interviews and recordings | Turn an audio file into text
Software | Developers | Add speech to text to an app | Run it through a Transformers pipeline
Education | Teachers and students | Caption lectures and talks | Get timestamps for each sentence
Skip it if you need the best accuracy, since the card says the speed gain costs a small drop in quality.
Where to use it:
Use case | Who | What they get back
Transcription | Anyone with recordings | The spoken words as text
Translation to English | Multilingual teams | English text from speech in another language
Timestamps | Caption makers | Time marks for sentences or for single words
How to set it up:
pip install --upgrade pip
pip install --upgrade transformers datasets[audio] accelerate
How to use it: Create a speech recognition pipeline, then call it with the path of your audio file. A list of files with a batch size lets it transcribe several at once.
import torch
device = "cuda:0" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
model_id = "openai/whisper-large-v3-turbo"
model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
)
model.to(device)
processor = AutoProcessor.from_pretrained(model_id)
pipe = pipeline(
"automatic-speech-recognition",
model=model,
tokenizer=processor.tokenizer,
feature_extractor=processor.feature_extractor,
torch_dtype=torch_dtype,
device=device,
)
result = pipe("audio.mp3")
print(result["text"])
Put your own value in its key setting first.
Option | What it does
pipe("audio.mp3") | Transcribes one local file
pipe(list, batch_size=2) | Transcribes several files in parallel
language in generate_kwargs | Tells the model the language when you already know it
task set to translate | Gives English text from speech in another language
return_timestamps=True | Adds sentence-level timestamps
return_timestamps="word" | Adds word-level timestamps
Title: Decoding layers
Whisper large-v3: 32 layers
Whisper large-v3-turbo: 4 layers
Example: You pass the path of an interview recording to the pipeline and print the text. Add the word-level timestamp option and you also get a time mark for each word.
For newcomers and experts: If you are new to speech recognition, run the file example first and check the text against what you hear. Then try timestamps.
If you know the field, the card says Transformers supports the Whisper decoding strategies, including temperature fallback and conditioning on previous tokens. Whisper predicts the language itself unless you pass one.
Install Transformers and friends
Load the model and processor
Build a speech recognition pipeline
Pass an audio file and read the text
Good to know:
Download its files with Hugging Face's command line:
pip install -U huggingface_hub
hf download openai/whisper-large-v3-turbo
Then follow the usage in its model card. Nothing in this kit runs by itself.
README.md 6 KB
Listed Hugging Face models in the same category that share the most tags with this one.
| Product | Downloads, last 30 days | Likes | Parameters | License | Last change |
|---|---|---|---|---|---|
| whisper-large-v3-turbo This page | 4,969,987 | 3,524 | 809M | mit | |
| Distil-Whisper distil-large-v3Shares whisper · automatic-speech-recognition · audio | 754,155 | 386 | 756M | mit | |
| whisper-persian-v4Shares whisper · automatic-speech-recognition · transformers | 1,071 | 21 | 1.5B | apache-2.0 | |
| whisper-base-ar-quranShares whisper · automatic-speech-recognition · transformers | 353,756 | 195 | — | apache-2.0 | |
| whisperkit-coremlShares whisper · automatic-speech-recognition | 60,717,190 | 266 | — | mit | |
| AADDShares whisper · audio | 73 | 9 | — | mit |
Posts and comments that name this project on Hacker News, Bluesky, X, Reddit, Facebook and YouTube, found by a weekly search and on Trendshift. Mindket doesn’t write or check them. Open one to read it where it was posted.
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