anton-l HF staff nielsr HF staff commited on
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27aba26
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Fix code snippets (#1)

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- Fix code snippets (0b10cd83fe40ae85538caf256570608dcb04c2e4)


Co-authored-by: Niels Rogge <nielsr@users.noreply.huggingface.co>

Files changed (1) hide show
  1. README.md +4 -4
README.md CHANGED
@@ -73,13 +73,13 @@ 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, Data2VecForCTC
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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-large-960h")
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- model = Data2VecForCTC.from_pretrained("facebook/data2vec-audio-large-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")
@@ -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-large-960h** on LibriSpeech's "clean" and "other" test data.
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  ```python
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- from transformers import Wav2Vec2Processor, Data2VecForCTC
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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-large-960h").to("cuda")
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- model = Data2VecForCTC.from_pretrained("facebook/data2vec-audio-large-960h")
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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-large-960h")
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+ model = Data2VecAudioForCTC.from_pretrained("facebook/data2vec-audio-large-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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  This code snippet shows how to evaluate **facebook/data2vec-audio-large-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-large-960h").to("cuda")
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+ model = Data2VecAudioForCTC.from_pretrained("facebook/data2vec-audio-large-960h")
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  librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")