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@@ -27,7 +27,7 @@ model-index:
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  value: 59.46
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  ---
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- # Wav2Vec2-Large-XLSR-53-{language} Vietnamese
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  Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Vietnamese using the [Common Voice](https://huggingface.co/datasets/common_voice), and [FOSD](https://data.mendeley.com/datasets/k9sxg2twv4/4).
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  When using this model, make sure that your speech input is sampled at 16kHz.
@@ -51,15 +51,15 @@ 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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@@ -70,7 +70,7 @@ print("Reference:", test_dataset["sentence"][:2])
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  ## Evaluation
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- The model can be evaluated as follows on the {language} test data of Common Voice. # TODO: replace #TODO: replace language with your {language}, *e.g.* French
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  ```python
@@ -87,30 +87,30 @@ processor = Wav2Vec2Processor.from_pretrained("Nhut/wav2vec2-large-xlsr-vietname
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  model = Wav2Vec2ForCTC.from_pretrained("Nhut/wav2vec2-large-xlsr-vietnamese")
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  model.to("cuda")
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- chars_to_ignore_regex = '[\\\\\\+\\@\\ǀ\\,\\?\\.\\!\\-\\;\\:\\"\\“\\%\\‘\\”\\�0123456789]'
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  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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- \tbatch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
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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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  # Preprocessing the datasets.
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  # We need to read the aduio files as arrays
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  def evaluate(batch):
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- \tinputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
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- \twith torch.no_grad():
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- \t\tlogits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
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- \tpred_ids = torch.argmax(logits, dim=-1)
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- \tbatch["pred_strings"] = processor.batch_decode(pred_ids)
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- \treturn batch
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  result = test_dataset.map(evaluate, batched=True, batch_size=8)
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  value: 59.46
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  ---
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+ # Wav2Vec2-Large-XLSR-53-Vietnamese
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  Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Vietnamese using the [Common Voice](https://huggingface.co/datasets/common_voice), and [FOSD](https://data.mendeley.com/datasets/k9sxg2twv4/4).
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  When using this model, make sure that your speech input is sampled at 16kHz.
 
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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 Vietnamese test data of Common Voice.
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  ```python
 
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  model = Wav2Vec2ForCTC.from_pretrained("Nhut/wav2vec2-large-xlsr-vietnamese")
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  model.to("cuda")
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+ chars_to_ignore_regex = '[\\\\\\\\\\\\+\\\\@\\\\ǀ\\\\,\\\\?\\\\.\\\\!\\\\-\\\\;\\\\:\\\\"\\\\“\\\\%\\\\‘\\\\”\\\\�0123456789]'
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  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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+ \\tbatch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
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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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  # Preprocessing the datasets.
104
  # We need to read the aduio files as arrays
105
  def evaluate(batch):
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+ \\tinputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
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+ \\twith torch.no_grad():
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+ \\t\\tlogits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
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+ \\tpred_ids = torch.argmax(logits, dim=-1)
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+ \\tbatch["pred_strings"] = processor.batch_decode(pred_ids)
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+ \\treturn batch
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  result = test_dataset.map(evaluate, batched=True, batch_size=8)
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