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Update README.md
Browse filesFix Typo, update class name from `Data2VecForCTC` to `Data2VecAudioForCTC`
README.md
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@@ -73,19 +73,19 @@ For more information, please take a look at the [official paper](https://arxiv.o
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To transcribe audio files the model can be used as a standalone acoustic model as follows:
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```python
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from transformers import Wav2Vec2Processor,
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from datasets import load_dataset
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import torch
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# load model and processor
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processor = Wav2Vec2Processor.from_pretrained("facebook/data2vec-audio-base-960h")
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model =
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# load dummy dataset and read soundfiles
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ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
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# tokenize
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input_values = processor(ds[0]["audio"]["array"]
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# retrieve logits
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logits = model(input_values).logits
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@@ -100,14 +100,14 @@ To transcribe audio files the model can be used as a standalone acoustic model a
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This code snippet shows how to evaluate **facebook/data2vec-audio-base-960h** on LibriSpeech's "clean" and "other" test data.
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```python
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from transformers import Wav2Vec2Processor,
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from datasets import load_dataset
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import torch
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from jiwer import wer
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# load model and processor
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processor = Wav2Vec2Processor.from_pretrained("facebook/data2vec-audio-base-960h").to("cuda")
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model =
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librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")
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To transcribe audio files the model can be used as a standalone acoustic model as follows:
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```python
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from transformers import Wav2Vec2Processor, Data2VecAudioForCTC
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from datasets import load_dataset
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import torch
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# load model and processor
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processor = Wav2Vec2Processor.from_pretrained("facebook/data2vec-audio-base-960h")
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model = Data2VecAudioForCTC.from_pretrained("facebook/data2vec-audio-base-960h")
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# load dummy dataset and read soundfiles
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ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
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# tokenize
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input_values = processor(ds[0]["audio"]["array"], return_tensors="pt", padding="longest").input_values # Batch size 1
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# retrieve logits
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logits = model(input_values).logits
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This code snippet shows how to evaluate **facebook/data2vec-audio-base-960h** on LibriSpeech's "clean" and "other" test data.
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```python
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from transformers import Wav2Vec2Processor, Data2VecAudioForCTC
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from datasets import load_dataset
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import torch
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from jiwer import wer
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# load model and processor
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processor = Wav2Vec2Processor.from_pretrained("facebook/data2vec-audio-base-960h").to("cuda")
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model = Data2VecAudioForCTC.from_pretrained("facebook/data2vec-audio-base-960h")
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librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")
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