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EliezerCarvalho/WhisperLv3-PT
WhisperLv3-PT is a automatic speech recognition model from EliezerCarvalho. Use it when you need speech turned into text. The card lists the license as mit.
[!NOTE] This repository contains the <a href = "https://huggingface.co/inesc-id/WhisperLv3-X-PT-All" WhisperLv3-X-PT-All</a model with the <b generationconfig </b file updated according to the <a href = "https://huggi…
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Updated Aug 8, 2026
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From the Hugging Face model README
[!NOTE] This repository contains the <a href = "https://huggingface.co/inesc-id/WhisperLv3-X-PT-All"> WhisperLv3-X-PT-All</a> model with the <b> generation_config </b> file updated according to the <a href = "https://huggingface.co/openai/whisper-large-v3"> whisper-large-v3</a> model. This update is necessary to avoid errors in certain customization parameters, such as beam_search, task, language, etc. This rep also contains WhisperLv3-X-PT-All quantized to 4 Bits and 8 Bits using BitsAndBytes. <br> <b> llama.cpp does not support quantization for ASR models. </b>
This model is a fine-tuned WhisperX variant of openai/whisper-large-v3, trained for Portuguese (pt) (European Portuguese, Brazilian and African and Asian Portuguese varieties) automatic speech recognition (ASR). From CAMÕES work.
openai/whisper-large-v3task="transcribe")This model was quantized using BitsAndBytes, allowing for a drastic reduction in memory, which enables inference on more modest systems.
| File | Quantization | Size |
|---|---|---|
| <a href = "https://huggingface.co/EliezerCarvalho/WhisperLv3-PT/tree/main/WhisperLv3-PT-All%208Bit">8BIT</a> | Q8 | 1.78 GB |
| <a href = "https://huggingface.co/EliezerCarvalho/WhisperLv3-PT/tree/main/WhisperLv3-PT-All%204Bit">4BIT_NF</a> | Q4 | 1.06 GB |
If you want to use Hugging Face transformers to generate text, you can do something like this.
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
from transformes.audio_utils import load_audio
MODEL = "EliezerCarvalho/WhisperLv3-PT"
PROCESSOR_ASR = AutoProcessor.from_pretrained (MODEL)
MODEL_ASR = AutoModelForSpeechSeq2Seq.from_pretrained (MODEL)
WAV = load_audio (path, sampling_rate = PROCESSOR_ASR.feature_extractor.sampling_rate)
input = PROCESSOR_ASR (wav, sampling_rate = PROCESSOR_ASR.feature_extractor.sampling_rate, return_tensors = "pt")
inference = MODEL_ASR.generate (input["input_features"], return_timestamps = True, task = "transcribe", language = "pt", return_segments = True)