patrickvonplaten
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Update README.md
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README.md
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@@ -28,6 +28,7 @@ The only change from the existing ASR pipeline will be:
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```diff
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import torch
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-from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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+from transformers import Wav2Vec2ForCTC, Wav2Vec2ProcessorWithLM
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from datasets import load_dataset
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@@ -36,11 +37,13 @@ ds = load_dataset("common_voice", "es", split="test", streaming=True)
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sample = next(iter(ds))
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model = Wav2Vec2ForCTC.from_pretrained("patrickvonplaten/wav2vec2-large-xlsr-53-spanish-with-lm")
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-processor = Wav2Vec2Processor.from_pretrained("patrickvonplaten/wav2vec2-large-xlsr-53-spanish-with-lm")
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+processor = Wav2Vec2ProcessorWithLM.from_pretrained("patrickvonplaten/wav2vec2-large-xlsr-53-spanish-with-lm")
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-
input_values = processor(
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with torch.no_grad():
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logits = model(input_values).logits
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```diff
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import torch
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import torchaudio.functional as F
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-from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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+from transformers import Wav2Vec2ForCTC, Wav2Vec2ProcessorWithLM
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from datasets import load_dataset
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sample = next(iter(ds))
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resampled_audio = F.resample(torch.tensor(sample["audio"]["array"]), 48_000, 16_000).n
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model = Wav2Vec2ForCTC.from_pretrained("patrickvonplaten/wav2vec2-large-xlsr-53-spanish-with-lm")
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-processor = Wav2Vec2Processor.from_pretrained("patrickvonplaten/wav2vec2-large-xlsr-53-spanish-with-lm")
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+processor = Wav2Vec2ProcessorWithLM.from_pretrained("patrickvonplaten/wav2vec2-large-xlsr-53-spanish-with-lm")
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input_values = processor(resampled_audio, return_tensors="pt").input_values
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with torch.no_grad():
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logits = model(input_values).logits
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