142 lines
5.1 KiB
Markdown
142 lines
5.1 KiB
Markdown
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<!--Copyright 2021 The HuggingFace Team. All rights reserved.
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Licensed under the Apache License, Version 2.0 (the "License"); you may not use this file except in compliance with
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the License. You may obtain a copy of the License at
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http://www.apache.org/licenses/LICENSE-2.0
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Unless required by applicable law or agreed to in writing, software distributed under the License is distributed on
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an "AS IS" BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. See the License for the
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specific language governing permissions and limitations under the License.
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⚠️ Note that this file is in Markdown but contain specific syntax for our doc-builder (similar to MDX) that may not be
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rendered properly in your Markdown viewer.
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-->
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*This model was released on 2021-06-14 and added to Hugging Face Transformers on 2021-06-16.*
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<div style="float: right;">
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<div class="flex flex-wrap space-x-1">
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<img alt="PyTorch" src="https://img.shields.io/badge/PyTorch-DE3412?style=flat&logo=pytorch&logoColor=white">
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<img alt="FlashAttention" src="https://img.shields.io/badge/%E2%9A%A1%EF%B8%8E%20FlashAttention-eae0c8?style=flat">
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<img alt="SDPA" src="https://img.shields.io/badge/SDPA-DE3412?style=flat&logo=pytorch&logoColor=white">
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</div>
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</div>
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# HuBERT
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[HuBERT](https://huggingface.co/papers/2106.07447) is a self-supervised speech model to cluster aligned target labels for BERT-like prediction loss and applying the prediction loss only over masked regions to force the model to learn both acoustic and language modeling over continuous inputs. It addresses the challenges of multiple sound units per utterance, no lexicon during pre-training, and variable-length sound units without explicit segmentation.
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You can find all the original HuBERT checkpoints under the [HuBERT](https://huggingface.co/collections/facebook/hubert-651fca95d57549832161e6b6) collection.
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> [!TIP]
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> This model was contributed by [patrickvonplaten](https://huggingface.co/patrickvonplaten).
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>
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> Click on the HuBERT models in the right sidebar for more examples of how to apply HuBERT to different audio tasks.
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The example below demonstrates how to automatically transcribe speech into text with [`Pipeline`] or the [`AutoModel`] class.
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<hfoptions id="usage">
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<hfoption id="Pipeline">
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```python
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import torch
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from transformers import pipeline
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pipeline = pipeline(
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task="automatic-speech-recognition",
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model="facebook/hubert-large-ls960-ft",
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dtype=torch.float16,
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device=0
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)
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pipeline("https://huggingface.co/datasets/Narsil/asr_dummy/resolve/main/1.flac")
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```
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</hfoption>
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<hfoption id="AutoModel">
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```python
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import torch
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from transformers import AutoProcessor, AutoModelForCTC
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from datasets import load_dataset
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dataset = load_dataset("hf-internal-testing/librispeech_asr_demo", "clean", split="validation").sort("id")
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sampling_rate = dataset.features["audio"].sampling_rate
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processor = AutoProcessor.from_pretrained("facebook/hubert-base-ls960")
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model = AutoModelForCTC.from_pretrained("facebook/hubert-base-ls960", dtype=torch.float16, device_map="auto", attn_implementation="sdpa")
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inputs = processor(dataset[0]["audio"]["array"], sampling_rate=sampling_rate, return_tensors="pt")
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with torch.no_grad():
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logits = model(**inputs).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription = processor.batch_decode(predicted_ids)
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print(transcription[0])
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```
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</hfoption>
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</hfoptions>
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## Quantization
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Quantization reduces the memory burden of large models by representing the weights in a lower precision.
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Refer to the [Quantization](../quantization/overview) overview for more available quantization backends.
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The example below uses [bitsandbytes](../quantization/bitsandbytes) to quantize the weights to 4-bits.
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```python
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import torch
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from transformers import AutoProcessor, AutoModelForCTC, BitsAndBytesConfig
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from datasets import load_dataset
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bnb_config = BitsAndBytesConfig(
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load_in_8bit=True,
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llm_int8_threshold=6.0
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)
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dataset = load_dataset("hf-internal-testing/librispeech_asr_demo", "clean", split="validation").sort("id")
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sampling_rate = dataset.features["audio"].sampling_rate
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processor = AutoProcessor.from_pretrained("facebook/hubert-base-ls960")
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model = AutoModelForCTC.from_pretrained("facebook/hubert-base-ls960", quantization_config=bnb_config, dtype=torch.float16, device_map="auto", attn_implementation="sdpa")
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inputs = processor(dataset[0]["audio"]["array"], sampling_rate=sampling_rate, return_tensors="pt")
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with torch.no_grad():
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logits = model(**inputs).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription = processor.batch_decode(predicted_ids)
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print(transcription[0])
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```
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## Notes
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- HuBERT models expect raw audio input as a 1D float array sampled at 16kHz.
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- If you want to use a `head_mask`, use the model with `attn_implementation="eager"`.
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```python
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model = HubertModel.from_pretrained("facebook/hubert-base-ls960", attn_implementation="eager")
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```
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## HubertConfig
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[[autodoc]] HubertConfig
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- all
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## HubertModel
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[[autodoc]] HubertModel
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- forward
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## HubertForCTC
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[[autodoc]] HubertForCTC
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- forward
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## HubertForSequenceClassification
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[[autodoc]] HubertForSequenceClassification
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- forward
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