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Update README.md

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Patrick von Platen 2022-03-31 19:51:10 +00:00 committed by huggingface-web
parent 63fb1ed2d0
commit d72d8ffa1e
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@ -48,25 +48,17 @@ To transcribe audio files the model can be used as a standalone acoustic model a
```python ```python
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
from datasets import load_dataset from datasets import load_dataset
import soundfile as sf
import torch import torch
# load model and processor # load model and processor
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-large-960h-lv60-self") processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-large-960h-lv60-self")
model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h-lv60-self") model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h-lv60-self")
# define function to read in sound file
def map_to_array(batch):
speech, _ = sf.read(batch["file"])
batch["speech"] = speech
return batch
# load dummy dataset and read soundfiles # load dummy dataset and read soundfiles
ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation") ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
ds = ds.map(map_to_array)
# tokenize # tokenize
input_values = processor(ds["speech"][:2], return_tensors="pt", padding="longest").input_values # Batch size 1 input_values = processor(batch["audio"]["array"], return_tensors="pt", padding="longest").input_values
# retrieve logits # retrieve logits
logits = model(input_values).logits logits = model(input_values).logits
@ -83,7 +75,6 @@ To transcribe audio files the model can be used as a standalone acoustic model a
```python ```python
from datasets import load_dataset from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import soundfile as sf
import torch import torch
from jiwer import wer from jiwer import wer
@ -93,15 +84,8 @@ librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")
model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h-lv60-self").to("cuda") model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h-lv60-self").to("cuda")
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-large-960h-lv60-self") processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-large-960h-lv60-self")
def map_to_array(batch):
speech, _ = sf.read(batch["file"])
batch["speech"] = speech
return batch
librispeech_eval = librispeech_eval.map(map_to_array)
def map_to_pred(batch): def map_to_pred(batch):
inputs = processor(batch["speech"], return_tensors="pt", padding="longest") inputs = processor(batch["audio"]["array"], return_tensors="pt", padding="longest")
input_values = inputs.input_values.to("cuda") input_values = inputs.input_values.to("cuda")
attention_mask = inputs.attention_mask.to("cuda") attention_mask = inputs.attention_mask.to("cuda")
@ -113,7 +97,7 @@ def map_to_pred(batch):
batch["transcription"] = transcription batch["transcription"] = transcription
return batch return batch
result = librispeech_eval.map(map_to_pred, batched=True, batch_size=16, remove_columns=["speech"]) result = librispeech_eval.map(map_to_pred, remove_columns=["speech"])
print("WER:", wer(result["text"], result["transcription"])) print("WER:", wer(result["text"], result["transcription"]))
``` ```