Automatic Speech Recognition
Transformers
PyTorch
JAX
Kyrgyz
wav2vec2
audio
speech
xlsr-fine-tuning-week
Eval Results (legacy)
Instructions to use iarfmoose/wav2vec2-large-xlsr-kyrgyz with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use iarfmoose/wav2vec2-large-xlsr-kyrgyz with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="iarfmoose/wav2vec2-large-xlsr-kyrgyz")# Load model directly from transformers import AutoProcessor, AutoModelForCTC processor = AutoProcessor.from_pretrained("iarfmoose/wav2vec2-large-xlsr-kyrgyz") model = AutoModelForCTC.from_pretrained("iarfmoose/wav2vec2-large-xlsr-kyrgyz", device_map="auto") - Notebooks
- Google Colab
- Kaggle
Adam Montgomerie commited on
Commit ·
dc7d85c
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Parent(s): 05f272c
Update README.md
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README.md
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@@ -49,15 +49,15 @@ resampler = torchaudio.transforms.Resample(48_000, 16_000)
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# Preprocessing the datasets.
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# We need to read the aduio files as arrays
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def speech_file_to_array_fn(batch):
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\tspeech_array, sampling_rate = torchaudio.load(batch["path"])
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\tbatch["speech"] = resampler(speech_array).squeeze().numpy()
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\treturn batch
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test_dataset = test_dataset.map(speech_file_to_array_fn)
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inputs = processor(test_dataset["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)
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with torch.no_grad():
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\tlogits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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## Evaluation
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The model can be evaluated as follows on the
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```python
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model = Wav2Vec2ForCTC.from_pretrained("iarfmoose/wav2vec2-large-xlsr-kyrgyz")
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model.to("cuda")
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chars_to_ignore_regex = '[\\,\\?\\.\\!\\-\\;\\:\\"\\“\\%\\‘\\”\\�\\–\\—\\¬\\⅛]'
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resampler = torchaudio.transforms.Resample(48_000, 16_000)
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def speech_file_to_array_fn(batch):
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# Preprocessing the datasets.
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# We need to read the aduio files as arrays
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def speech_file_to_array_fn(batch):
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\\tspeech_array, sampling_rate = torchaudio.load(batch["path"])
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\\tbatch["speech"] = resampler(speech_array).squeeze().numpy()
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\\treturn batch
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test_dataset = test_dataset.map(speech_file_to_array_fn)
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inputs = processor(test_dataset["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)
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with torch.no_grad():
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\\tlogits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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## Evaluation
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The model can be evaluated as follows on the Kyrgyz test data of Common Voice.
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```python
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model = Wav2Vec2ForCTC.from_pretrained("iarfmoose/wav2vec2-large-xlsr-kyrgyz")
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model.to("cuda")
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chars_to_ignore_regex = '[\\\\,\\\\?\\\\.\\\\!\\\\-\\\\;\\\\:\\\\"\\\\“\\\\%\\\\‘\\\\”\\\\�\\\\–\\\\—\\\\¬\\\\⅛]'
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resampler = torchaudio.transforms.Resample(48_000, 16_000)
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def speech_file_to_array_fn(batch):
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