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--- |
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language: zh |
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datasets: |
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- common_voice |
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tags: |
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- speech |
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- audio |
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- automatic-speech-recognition |
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- xlsr_fine_tuning_week |
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license: apache-2.0 |
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--- |
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## Colab trial with recording or voice file |
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[Colab trial](https://colab.research.google.com/drive/1nBRLf4Pwiply_y5rXWoaIB8LxX41tfEI?usp=sharing) |
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``` |
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import torchaudio |
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from datasets import load_dataset, load_metric |
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from transformers import ( |
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Wav2Vec2ForCTC, |
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Wav2Vec2Processor, |
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) |
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import torch |
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import re |
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import sys |
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model_name = "voidful/wav2vec2-large-xlsr-53-hk" |
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device = "cuda" |
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processor_name = "voidful/wav2vec2-large-xlsr-53-hk" |
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chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\"#$%&()*+,\-.\:;<=>?@\[\]\\\/^_`{|}~]" |
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model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device) |
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processor = Wav2Vec2Processor.from_pretrained(processor_name) |
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resampler = torchaudio.transforms.Resample(orig_freq=48_000, new_freq=16_000) |
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def load_file_to_data(file): |
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batch = {} |
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speech, _ = torchaudio.load(file) |
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batch["speech"] = resampler.forward(speech.squeeze(0)).numpy() |
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batch["sampling_rate"] = resampler.new_freq |
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return batch |
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def predict(data): |
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features = processor(data["speech"], sampling_rate=data["sampling_rate"], padding=True, return_tensors="pt") |
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input_values = features.input_values.to(device) |
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attention_mask = features.attention_mask.to(device) |
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with torch.no_grad(): |
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logits = model(input_values, attention_mask=attention_mask).logits |
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pred_ids = torch.argmax(logits, dim=-1) |
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return processor.batch_decode(pred_ids) |
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``` |
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Predict |
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```python |
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predict(load_file_to_data('voice file path')) |
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``` |
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## Evaluation on Common Voice HK Test |
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```python |
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import torchaudio |
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from datasets import load_dataset, load_metric |
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from transformers import ( |
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Wav2Vec2ForCTC, |
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Wav2Vec2Processor, |
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) |
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import torch |
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import re |
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import sys |
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|
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model_name = "voidful/wav2vec2-large-xlsr-53-hk" |
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device = "cuda" |
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processor_name = "voidful/wav2vec2-large-xlsr-53-hk" |
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chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\"#$%&()*+,\-.\:;<=>?@\[\]\\\/^_`{|}~]" |
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model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device) |
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processor = Wav2Vec2Processor.from_pretrained(processor_name) |
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ds = load_dataset("common_voice", 'zh-HK', data_dir="./cv-corpus-6.1-2020-12-11", split="test") |
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resampler = torchaudio.transforms.Resample(orig_freq=48_000, new_freq=16_000) |
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def map_to_array(batch): |
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speech, _ = torchaudio.load(batch["path"]) |
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batch["speech"] = resampler.forward(speech.squeeze(0)).numpy() |
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batch["sampling_rate"] = resampler.new_freq |
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batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'") |
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return batch |
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ds = ds.map(map_to_array) |
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def map_to_pred(batch): |
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features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt") |
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input_values = features.input_values.to(device) |
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attention_mask = features.attention_mask.to(device) |
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with torch.no_grad(): |
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logits = model(input_values, attention_mask=attention_mask).logits |
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pred_ids = torch.argmax(logits, dim=-1) |
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batch["predicted"] = processor.batch_decode(pred_ids) |
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batch["target"] = batch["sentence"] |
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return batch |
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result = ds.map(map_to_pred, batched=True, batch_size=16, remove_columns=list(ds.features.keys())) |
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wer = load_metric("wer") |
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print(wer.compute(predictions=result["predicted"], references=result["target"])) |
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``` |