anaksor-ocr
Sokheng · Kit
Reads Khmer and English text from documents, receipts and photos of signs, using a detector for text lines and a recogniser for each line. Suits developers who…
Turns English speech into text and runs about 6.3 times faster than Whisper large-v3 while staying close in accuracy on long audio. For developers and teams who transcribe recordings.
Distil-Whisper distil-large-v3 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 accelerate datasets[audio]
distil-large-v3 is an English speech recognition model. It is a distilled, smaller version of OpenAI's Whisper large-v3, with 756 million parameters against 1550 million. It was tuned for accurate transcription of long recordings.
The card says it stays within 1% word error rate of large-v3 on long audio and runs 6.3 times faster. It works with Transformers and with the popular Whisper libraries.
Developers and teams who need fast English transcription on their own hardware, for clips and for long recordings alike.
| Industry | Who uses it | How they use it | Example |
|---|---|---|---|
| Media | Podcast and video teams | Transcribe recordings to make captions and notes | Turning an interview recording into text |
| Software | App developers | Add speech to text inside a product | A voice note feature in an app |
| Education | Lecturers and students | Transcribe lectures for search and review | Writing up a recorded class |
| Research | Speech researchers | Compare distilled and full Whisper models | Measuring error rates on their own audio |
Anyone who needs transcription in languages other than English can skip it, since the card lists English only.
| Use case | Who | What they get back |
|---|---|---|
| Short clip transcription | App developers | Text for audio under 30 seconds |
| Long recording transcription | Media teams | Text for long audio using sequential or chunked methods |
| Faster pipelines | Engineers | A smaller model that replaces large-v3 with a speed gain |
You need Python and a recent Transformers library.
pip install --upgrade pip pip install --upgrade transformers accelerate datasets[audio]
Load the model into a speech recognition pipeline, then pass it audio. This example transcribes a clip shorter than 30 seconds from a small test dataset.
import torch
from datasets import load_dataset
device = "cuda:0" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
model_id = "distil-whisper/distil-large-v3"
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,
max_new_tokens=128,
torch_dtype=torch_dtype,
device=device,
)
dataset = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
sample = dataset[0]["audio"]
result = pipe(sample)
print(result["text"])Put your own value in its key setting first.
To transcribe your own file, pass its path to the pipeline in place of the sample:
result = pipe("audio.mp3")| Tool | What it does |
|---|---|
| pipeline | Wraps the model so you can pass audio and get text back |
| Short-form | Handles clips shorter than 30 seconds |
| Sequential long-form | Transcribes longer audio with OpenAI's sequential algorithm |
| Chunked long-form | Splits long audio into chunks and joins the text |
Pass the path of a recording called audio.mp3 to the pipeline. The result holds the transcribed text under its text key, which you can print or save.
If you are new, a speech recognition model listens to audio and writes down the words. Install Transformers, run the short example once on the test clip, then swap in one of your own audio files.
For experts, the model was distilled so that the sequential long-form algorithm, which Whisper libraries use by default, gives accurate output. The card reports long-form error rates of 10.8 sequential and 10.9 chunked against 10.0 and 11.0 for large-v3. It beats distil-large-v2 on sequential long-form, at 10.8 against 15.6.
The points below cover limits and costs.
Turns English speech into text and runs about 6.3 times faster than Whisper large-v3 while staying close in accuracy on long audio. For developers and teams who transcribe recordings.
Made by distil-whisper 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: distil-large-v3 is an English speech recognition model. It is a distilled, smaller version of OpenAI's Whisper large-v3, with 756 million parameters against 1550 million. It was tuned for accurate transcription of long recordings.
The card says it stays within 1% word error rate of large-v3 on long audio and runs 6.3 times faster. It works with Transformers and with the popular Whisper libraries.
Who it suits: Developers and teams who need fast English transcription on their own hardware, for clips and for long recordings alike.
Industry | Who uses it | How they use it | Example
Media | Podcast and video teams | Transcribe recordings to make captions and notes | Turning an interview recording into text
Software | App developers | Add speech to text inside a product | A voice note feature in an app
Education | Lecturers and students | Transcribe lectures for search and review | Writing up a recorded class
Research | Speech researchers | Compare distilled and full Whisper models | Measuring error rates on their own audio
Anyone who needs transcription in languages other than English can skip it, since the card lists English only.
Where to use it:
Use case | Who | What they get back
Short clip transcription | App developers | Text for audio under 30 seconds
Long recording transcription | Media teams | Text for long audio using sequential or chunked methods
Faster pipelines | Engineers | A smaller model that replaces large-v3 with a speed gain
How to set it up: You need Python and a recent Transformers library.
pip install --upgrade pip
pip install --upgrade transformers accelerate datasets[audio]
How to use it: Load the model into a speech recognition pipeline, then pass it audio. This example transcribes a clip shorter than 30 seconds from a small test dataset.
import torch
from datasets import load_dataset
device = "cuda:0" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
model_id = "distil-whisper/distil-large-v3"
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,
max_new_tokens=128,
torch_dtype=torch_dtype,
device=device,
)
dataset = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
sample = dataset[0]["audio"]
result = pipe(sample)
print(result["text"])
Put your own value in its key setting first.
To transcribe your own file, pass its path to the pipeline in place of the sample:
result = pipe("audio.mp3")
Tool | What it does
pipeline | Wraps the model so you can pass audio and get text back
Short-form | Handles clips shorter than 30 seconds
Sequential long-form | Transcribes longer audio with OpenAI's sequential algorithm
Chunked long-form | Splits long audio into chunks and joins the text
Example: Pass the path of a recording called audio.mp3 to the pipeline. The result holds the transcribed text under its text key, which you can print or save.
For newcomers and experts: If you are new, a speech recognition model listens to audio and writes down the words. Install Transformers, run the short example once on the test clip, then swap in one of your own audio files.
For experts, the model was distilled so that the sequential long-form algorithm, which Whisper libraries use by default, gives accurate output. The card reports long-form error rates of 10.8 sequential and 10.9 chunked against 10.0 and 11.0 for large-v3. It beats distil-large-v2 on sequential long-form, at 10.8 against 15.6.
Install Transformers and its helpers
Load the model and processor
Build a speech recognition pipeline
Pass an audio clip or file path
Read the transcribed text from the result
Title: Speed compared with large-v3, higher is faster
large-v3: 1.0 times
distil-large-v2: 5.8 times
distil-large-v3: 6.3 times
Good to know: The points below cover limits and costs.
Download its files with Hugging Face's command line:
pip install -U huggingface_hub
hf download distil-whisper/distil-large-v3
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 |
|---|---|---|---|---|---|
| Distil-Whisper distil-large-v3 This page | 754,155 | 386 | 756M | mit | |
| whisper-persian-v4Shares tensorboard · whisper · automatic-speech-recognition | 1,071 | 21 | 1.5B | apache-2.0 | |
| whisper-base-ar-quranShares tensorboard · whisper · automatic-speech-recognition | 353,756 | 195 | — | apache-2.0 | |
| SmolLM2-135M-InstructShares tensorboard · transformers.js · onnx | 841,106 | 474 | 135M | apache-2.0 | |
| SmolLM2-1.7B-InstructShares tensorboard · transformers.js · onnx | 267,519 | 772 | 1.7B | apache-2.0 | |
| SmolLM2-360M-InstructShares tensorboard · transformers.js · onnx | 221,273 | 244 | 362M | apache-2.0 |
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