Update README.md
Browse files
README.md
CHANGED
|
@@ -28,7 +28,7 @@ model-index:
|
|
| 28 |
|
| 29 |
# Wav2Vec2-Large-XLSR-53-Tamil
|
| 30 |
|
| 31 |
-
Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Tamil using the [Common Voice](https://huggingface.co/datasets/common_voice)
|
| 32 |
When using this model, make sure that your speech input is sampled at 16kHz.
|
| 33 |
|
| 34 |
## Usage
|
|
@@ -47,13 +47,13 @@ resampler = torchaudio.transforms.Resample(48_000, 16_000)
|
|
| 47 |
# Preprocessing the datasets.
|
| 48 |
# We need to read the aduio files as arrays
|
| 49 |
def speech_file_to_array_fn(batch):
|
| 50 |
-
|
| 51 |
-
|
| 52 |
-
|
| 53 |
test_dataset = test_dataset.map(speech_file_to_array_fn)
|
| 54 |
inputs = processor(test_dataset["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)
|
| 55 |
with torch.no_grad():
|
| 56 |
-
|
| 57 |
predicted_ids = torch.argmax(logits, dim=-1)
|
| 58 |
print("Prediction:", processor.batch_decode(predicted_ids))
|
| 59 |
print("Reference:", test_dataset["sentence"][:2])
|
|
@@ -76,25 +76,25 @@ wer = load_metric("wer")
|
|
| 76 |
processor = Wav2Vec2Processor.from_pretrained("{model_id}") #TODO: replace {model_id} with your model id. The model id consists of {your_username}/{your_modelname}, *e.g.* `elgeish/wav2vec2-large-xlsr-53-arabic`
|
| 77 |
model = Wav2Vec2ForCTC.from_pretrained("{model_id}") #TODO: replace {model_id} with your model id. The model id consists of {your_username}/{your_modelname}, *e.g.* `elgeish/wav2vec2-large-xlsr-53-arabic`
|
| 78 |
model.to("cuda")
|
| 79 |
-
chars_to_ignore_regex = '[
|
| 80 |
resampler = torchaudio.transforms.Resample(48_000, 16_000)
|
| 81 |
# Preprocessing the datasets.
|
| 82 |
# We need to read the audio files as arrays
|
| 83 |
def speech_file_to_array_fn(batch):
|
| 84 |
-
|
| 85 |
-
|
| 86 |
-
|
| 87 |
-
|
| 88 |
test_dataset = test_dataset.map(speech_file_to_array_fn)
|
| 89 |
# Preprocessing the datasets.
|
| 90 |
# We need to read the aduio files as arrays
|
| 91 |
def evaluate(batch):
|
| 92 |
-
|
| 93 |
-
|
| 94 |
-
|
| 95 |
-
|
| 96 |
-
|
| 97 |
-
|
| 98 |
result = test_dataset.map(evaluate, batched=True, batch_size=8)
|
| 99 |
print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
|
| 100 |
```
|
|
@@ -104,6 +104,6 @@ print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"],
|
|
| 104 |
|
| 105 |
## Training
|
| 106 |
|
| 107 |
-
The Common Voice `train`, `validation
|
| 108 |
|
| 109 |
-
The script used for training can be found [
|
|
|
|
| 28 |
|
| 29 |
# Wav2Vec2-Large-XLSR-53-Tamil
|
| 30 |
|
| 31 |
+
Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Tamil using the [Common Voice](https://huggingface.co/datasets/common_voice) dataset.
|
| 32 |
When using this model, make sure that your speech input is sampled at 16kHz.
|
| 33 |
|
| 34 |
## Usage
|
|
|
|
| 47 |
# Preprocessing the datasets.
|
| 48 |
# We need to read the aduio files as arrays
|
| 49 |
def speech_file_to_array_fn(batch):
|
| 50 |
+
\\tspeech_array, sampling_rate = torchaudio.load(batch["path"])
|
| 51 |
+
\\tbatch["speech"] = resampler(speech_array).squeeze().numpy()
|
| 52 |
+
\\treturn batch
|
| 53 |
test_dataset = test_dataset.map(speech_file_to_array_fn)
|
| 54 |
inputs = processor(test_dataset["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)
|
| 55 |
with torch.no_grad():
|
| 56 |
+
\\tlogits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
|
| 57 |
predicted_ids = torch.argmax(logits, dim=-1)
|
| 58 |
print("Prediction:", processor.batch_decode(predicted_ids))
|
| 59 |
print("Reference:", test_dataset["sentence"][:2])
|
|
|
|
| 76 |
processor = Wav2Vec2Processor.from_pretrained("{model_id}") #TODO: replace {model_id} with your model id. The model id consists of {your_username}/{your_modelname}, *e.g.* `elgeish/wav2vec2-large-xlsr-53-arabic`
|
| 77 |
model = Wav2Vec2ForCTC.from_pretrained("{model_id}") #TODO: replace {model_id} with your model id. The model id consists of {your_username}/{your_modelname}, *e.g.* `elgeish/wav2vec2-large-xlsr-53-arabic`
|
| 78 |
model.to("cuda")
|
| 79 |
+
chars_to_ignore_regex = '[\\\\,\\\\?\\\\.\\\\!\\\\-\\\\;\\\\:\\\\"\\\\“]' # TODO: adapt this list to include all special characters you removed from the data
|
| 80 |
resampler = torchaudio.transforms.Resample(48_000, 16_000)
|
| 81 |
# Preprocessing the datasets.
|
| 82 |
# We need to read the audio files as arrays
|
| 83 |
def speech_file_to_array_fn(batch):
|
| 84 |
+
\\tbatch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
|
| 85 |
+
\\tspeech_array, sampling_rate = torchaudio.load(batch["path"])
|
| 86 |
+
\\tbatch["speech"] = resampler(speech_array).squeeze().numpy()
|
| 87 |
+
\\treturn batch
|
| 88 |
test_dataset = test_dataset.map(speech_file_to_array_fn)
|
| 89 |
# Preprocessing the datasets.
|
| 90 |
# We need to read the aduio files as arrays
|
| 91 |
def evaluate(batch):
|
| 92 |
+
\\tinputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
|
| 93 |
+
\\twith torch.no_grad():
|
| 94 |
+
\\t\\tlogits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
|
| 95 |
+
\\tpred_ids = torch.argmax(logits, dim=-1)
|
| 96 |
+
\\tbatch["pred_strings"] = processor.batch_decode(pred_ids)
|
| 97 |
+
\\treturn batch
|
| 98 |
result = test_dataset.map(evaluate, batched=True, batch_size=8)
|
| 99 |
print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
|
| 100 |
```
|
|
|
|
| 104 |
|
| 105 |
## Training
|
| 106 |
|
| 107 |
+
The Common Voice `train`, `validation` were used for training
|
| 108 |
|
| 109 |
+
The script used for training can be found [https://colab.research.google.com/drive/1PC2SjxpcWMQ2qmRw21NbP38wtQQUa5os#scrollTo=YKBZdqqJG9Tv](...)
|