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Update README.md
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README.md
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@@ -31,13 +31,13 @@ The original model can be found under https://github.com/pytorch/fairseq/tree/ma
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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
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from datasets import load_dataset
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import soundfile as sf
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import torch
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# load model and
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model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h")
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# define function to read in sound file
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ds = ds.map(map_to_array)
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# tokenize
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input_values =
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# retrieve logits
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logits = model(input_values).logits
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# take argmax and decode
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription =
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```
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## Evaluation
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```python
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from datasets import load_dataset
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from transformers import Wav2Vec2ForCTC,
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import soundfile as sf
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import torch
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from jiwer import wer
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librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")
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model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h").to("cuda")
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def map_to_array(batch):
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speech, _ = sf.read(batch["file"])
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librispeech_eval = librispeech_eval.map(map_to_array)
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def map_to_pred(batch):
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input_values =
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with torch.no_grad():
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logits = model(input_values.to("cuda")).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription =
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batch["transcription"] = transcription
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return batch
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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, Wav2Vec2ForCTC
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from datasets import load_dataset
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import soundfile as sf
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import torch
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# load model and processor
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processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-large-960h")
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model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h")
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# define function to read in sound file
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ds = ds.map(map_to_array)
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# tokenize
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input_values = processor(ds["speech"][:2], 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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# take argmax and decode
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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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```
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## Evaluation
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```python
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from datasets import load_dataset
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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import soundfile as sf
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import torch
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from jiwer import wer
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librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")
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model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h").to("cuda")
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processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-large-960h")
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def map_to_array(batch):
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speech, _ = sf.read(batch["file"])
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librispeech_eval = librispeech_eval.map(map_to_array)
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def map_to_pred(batch):
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input_values = processor(batch["speech"], return_tensors="pt", padding="longest").input_values
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with torch.no_grad():
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logits = model(input_values.to("cuda")).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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batch["transcription"] = transcription
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return batch
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