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…
Turns speech in audio files into timestamped text, as OpenAI's Whisper large-v3 converted for CTranslate2 and faster-whisper. For developers who transcribe recordings in many languages on their own machine.
faster-whisper-large-v3 is a kit in the Productivity 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 -U huggingface_hub hf download Systran/faster-whisper-large-v3
This repository holds OpenAI's Whisper large-v3 speech recognition model, converted to the CTranslate2 format. You can use it in CTranslate2 or in projects built on it, such as faster-whisper. It transcribes audio to text with start and end times for each segment.
Developers, creators and researchers who want to transcribe audio locally and can run a Python script.
| Industry | Who uses it | How they use it | Example |
|---|---|---|---|
| Media | Podcasters and video editors | Transcribe recordings into text | Turn an interview into captions |
| Research | Interviewers and analysts | Convert spoken material into searchable text | Transcribe field recordings |
| Software | Developers | Add speech to text to an app | Build a transcription feature around faster-whisper |
If you only need a hosted transcription service with no setup, you can skip it.
| Use case | Who | What they get back |
|---|---|---|
| Transcription | Creators | Text for each audio segment |
| Captions | Video editors | Segment start and end times with text |
| Multilingual audio | Researchers | Transcripts across many languages |
pip install -U huggingface_hub hf download Systran/faster-whisper-large-v3
Load WhisperModel with the large-v3 name, call transcribe on an audio file, then loop over the segments and print each one with its times.
from faster_whisper import WhisperModel
model = WhisperModel("large-v3")
segments, info = model.transcribe("audio.mp3")
for segment in segments:
print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))You run the script on a file named audio.mp3. It prints one line per segment, each with a start time, an end time and the spoken text.
If you are new, run the script on a short clip first. The first run needs the model files, which are large.
For experts, the conversion used ct2-transformers-converter with float16 quantization, copying the tokenizer and preprocessor config. Weights are saved in FP16, and you can change that when loading through the compute_type option in CTranslate2.
The card shows the conversion command it used.
ct2-transformers-converter --model openai/whisper-large-v3 --output_dir faster-whisper-large-v3 \
--copy_files tokenizer.json preprocessor_config.json --quantization float16Turns speech in audio files into timestamped text, as OpenAI's Whisper large-v3 converted for CTranslate2 and faster-whisper. For developers who transcribe recordings in many languages on their own machine.
Made by Systran 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: This repository holds OpenAI's Whisper large-v3 speech recognition model, converted to the CTranslate2 format. You can use it in CTranslate2 or in projects built on it, such as faster-whisper. It transcribes audio to text with start and end times for each segment.
Who it suits: Developers, creators and researchers who want to transcribe audio locally and can run a Python script.
Industry | Who uses it | How they use it | Example
Media | Podcasters and video editors | Transcribe recordings into text | Turn an interview into captions
Research | Interviewers and analysts | Convert spoken material into searchable text | Transcribe field recordings
Software | Developers | Add speech to text to an app | Build a transcription feature around faster-whisper
If you only need a hosted transcription service with no setup, you can skip it.
Where to use it:
Use case | Who | What they get back
Transcription | Creators | Text for each audio segment
Captions | Video editors | Segment start and end times with text
Multilingual audio | Researchers | Transcripts across many languages
How to set it up:
pip install -U huggingface_hub
hf download Systran/faster-whisper-large-v3
How to use it: Load WhisperModel with the large-v3 name, call transcribe on an audio file, then loop over the segments and print each one with its times.
from faster_whisper import WhisperModel
model = WhisperModel("large-v3")
segments, info = model.transcribe("audio.mp3")
for segment in segments:
print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
Example: You run the script on a file named audio.mp3. It prints one line per segment, each with a start time, an end time and the spoken text.
For newcomers and experts: If you are new, run the script on a short clip first. The first run needs the model files, which are large.
For experts, the conversion used ct2-transformers-converter with float16 quantization, copying the tokenizer and preprocessor config. Weights are saved in FP16, and you can change that when loading through the compute_type option in CTranslate2.
OpenAI Whisper large-v3 is the source
ct2-transformers-converter converts it
CTranslate2 or faster-whisper loads it
You get timestamped text
Good to know: The card shows the conversion command it used.
ct2-transformers-converter --model openai/whisper-large-v3 --output_dir faster-whisper-large-v3 \
--copy_files tokenizer.json preprocessor_config.json --quantization float16
Download its files with Hugging Face's command line:
pip install -U huggingface_hub
hf download Systran/faster-whisper-large-v3
Then follow the usage in its model card. Nothing in this kit runs by itself.
README.md 4 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 |
|---|---|---|---|---|---|
| faster-whisper-large-v3 This page | 1,660,044 | 714 | — | mit | |
| whisper-large-v3Shares audio · automatic-speech-recognition | 3,489,398 | 6,802 | 1.5B | apache-2.0 | |
| Ming-3.0-audio-tinyShares audio · automatic-speech-recognition | 5,202 | 62 | 8.9B | mit | |
| Haas-ja-v1.0Shares audio · automatic-speech-recognition | 3,665 | 77 | — | apache-2.0 | |
| orukqwen-asr-1.7bShares audio · automatic-speech-recognition | 222 | 6 | 2B | apache-2.0 | |
| Mega-ASRShares audio · automatic-speech-recognition | 0 | 104 | — | apache-2.0 |
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