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
Commit ·
fb0e041
1
Parent(s): dc7d85c
Update README.md
Browse files
README.md
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@@ -17,7 +17,7 @@ model-index:
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dataset:
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name: Common Voice ky
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type: common_voice
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args:
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metrics:
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- name: Test WER
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type: wer
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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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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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dataset:
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name: Common Voice ky
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type: common_voice
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args: ky
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metrics:
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- name: Test WER
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type: wer
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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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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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