Commit
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4ec0637
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Parent(s):
6818035
Update README and model file
Browse files- 4gram.zip +2 -2
- README.md +38 -21
- example.mp3 +0 -0
- example.wav +0 -0
- example2.mp3 +0 -0
- hyperparams.yaml +1 -1
- model.ckpt +1 -1
4gram.zip
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README.md
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- Transformer
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license: cc-by-nc-4.0
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widget:
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- example_title:
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src: https://huggingface.co/
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- example_title:
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src: https://huggingface.co/
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- example_title: VLSP ASR 2020 test T2
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src: https://huggingface.co/nguyenvulebinh/wav2vec2-base-vietnamese-250h/raw/main/audio-test/t2_0000006682.wav
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model-index:
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- name: Wav2vec2 Base Vietnamese 270h
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results:
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- name: Test WER
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type: wer
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value: 9.66
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- task:
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name: Speech Recognition
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type: automatic-speech-recognition
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metrics:
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- name: Test WER
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type: wer
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value:
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---
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# Wav2Vec2-Base-Vietnamese-270h
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Fine-tuned Wav2Vec2 model on Vietnamese Speech Recognition task using about 270h labelled data combined from multiple datasets including [Common Voice](https://huggingface.co/datasets/common_voice), [VIVOS](https://huggingface.co/datasets/vivos), [VLSP2020](https://vlsp.org.vn/vlsp2020/eval/asr). The model was fine-tuned using SpeechBrain toolkit with a custom tokenizer. For a better experience, we encourage you to learn more about [SpeechBrain](https://speechbrain.github.io/).
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Please refer to [huggingface blog](https://huggingface.co/blog/fine-tune-wav2vec2-english) or [speechbrain](https://github.com/speechbrain/speechbrain/tree/develop/recipes/CommonVoice/ASR/CTC) on how to fine-tune Wav2Vec2 model on a specific language.
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### Benchmark WER result:
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| | [VIVOS](https://huggingface.co/datasets/vivos) | [COMMON VOICE
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|without LM| 8.
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|with 4-grams LM|
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The language model was trained using [OSCAR](https://huggingface.co/datasets/oscar-corpus/OSCAR-2109) dataset on about 32GB of crawled text.
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### Install SpeechBrain
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To use this model, you should install speechbrain
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```bash
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pip install git+https://github.com/speechbrain/speechbrain.git@develop
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```
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### Usage
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The model can be used directly (without a language model) as follows:
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from speechbrain.pretrained import EncoderASR
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model = EncoderASR.from_hparams(source="dragonSwing/wav2vec2-base-vn-270h", savedir="pretrained_models/asr-wav2vec2-vi")
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model.transcribe_file('dragonSwing/wav2vec2-base-vn-270h/example.
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```
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### Inference on GPU
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To perform inference on the GPU, add `run_opts={"device":"cuda"}` when calling the `from_hparams` method.
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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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import torch
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import torchaudio
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from transformers import Wav2Vec2FeatureExtractor
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from speechbrain.pretrained import EncoderASR
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import re
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test_dataset = load_dataset("
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test_dataset = test_dataset.cast_column("audio", Audio(sampling_rate=16_000))
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device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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wer = load_metric("wer")
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batch["pred_strings"] = pred_str
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return batch
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result = test_dataset.map(evaluate, batched=True, batch_size=
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print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["target_text"])))
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```
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**Test Result**:
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#### Citation
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```
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- Transformer
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license: cc-by-nc-4.0
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widget:
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- example_title: Example 1
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src: https://huggingface.co/dragonSwing/wav2vec2-base-vn-270h/raw/main/example.mp3
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- example_title: Example 2
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src: https://huggingface.co/dragonSwing/wav2vec2-base-vn-270h/raw/main/example2.mp3
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model-index:
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- name: Wav2vec2 Base Vietnamese 270h
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results:
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- name: Test WER
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type: wer
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value: 9.66
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- task:
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name: Speech Recognition
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type: automatic-speech-recognition
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dataset:
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name: Common Voice 7.0
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type: mozilla-foundation/common_voice_7_0
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args: vi
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metrics:
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- name: Test WER
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type: wer
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value: 5.57
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- task:
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name: Speech Recognition
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type: automatic-speech-recognition
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dataset:
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name: Common Voice 8.0
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type: mozilla-foundation/common_voice_8_0
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args: vi
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metrics:
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- name: Test WER
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type: wer
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value: 5.76
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- task:
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name: Speech Recognition
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type: automatic-speech-recognition
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metrics:
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- name: Test WER
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type: wer
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value: 3.70
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---
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# Wav2Vec2-Base-Vietnamese-270h
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Fine-tuned Wav2Vec2 model on Vietnamese Speech Recognition task using about 270h labelled data combined from multiple datasets including [Common Voice](https://huggingface.co/datasets/common_voice), [VIVOS](https://huggingface.co/datasets/vivos), [VLSP2020](https://vlsp.org.vn/vlsp2020/eval/asr). The model was fine-tuned using SpeechBrain toolkit with a custom tokenizer. For a better experience, we encourage you to learn more about [SpeechBrain](https://speechbrain.github.io/).
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Please refer to [huggingface blog](https://huggingface.co/blog/fine-tune-wav2vec2-english) or [speechbrain](https://github.com/speechbrain/speechbrain/tree/develop/recipes/CommonVoice/ASR/CTC) on how to fine-tune Wav2Vec2 model on a specific language.
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### Benchmark WER result:
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| | [VIVOS](https://huggingface.co/datasets/vivos) | [COMMON VOICE 7.0](https://huggingface.co/datasets/mozilla-foundation/common_voice_7_0) | [COMMON VOICE 8.0](https://huggingface.co/datasets/mozilla-foundation/common_voice_8_0) |
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|---|---|---|---|
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|without LM| 8.23 | 12.15 | 12.15 |
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|with 4-grams LM| 3.70 | 5.57 | 5.76 |
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The language model was trained using [OSCAR](https://huggingface.co/datasets/oscar-corpus/OSCAR-2109) dataset on about 32GB of crawled text.
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### Install SpeechBrain
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To use this model, you should install speechbrain > 0.5.10
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### Usage
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The model can be used directly (without a language model) as follows:
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from speechbrain.pretrained import EncoderASR
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model = EncoderASR.from_hparams(source="dragonSwing/wav2vec2-base-vn-270h", savedir="pretrained_models/asr-wav2vec2-vi")
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model.transcribe_file('dragonSwing/wav2vec2-base-vn-270h/example.mp3')
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# Output: được hồ chí minh coi là một động lực lớn của sự phát triển đất nước
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```
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### Inference on GPU
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To perform inference on the GPU, add `run_opts={"device":"cuda"}` when calling the `from_hparams` method.
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### Evaluation
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The model can be evaluated as follows on the Vietnamese test data of Common Voice 8.0.
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```python
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import torch
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import torchaudio
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from transformers import Wav2Vec2FeatureExtractor
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from speechbrain.pretrained import EncoderASR
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import re
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test_dataset = load_dataset("mozilla-foundation/common_voice_8_0", "vi", split="test", use_auth_token=True)
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test_dataset = test_dataset.cast_column("audio", Audio(sampling_rate=16_000))
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device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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wer = load_metric("wer")
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batch["pred_strings"] = pred_str
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return batch
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result = test_dataset.map(evaluate, batched=True, batch_size=1)
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print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["target_text"])))
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```
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**Test Result**: 12.155553%
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#### Citation
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```
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example.mp3
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example.wav
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example2.mp3
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hyperparams.yaml
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# ################################
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# Model: wav2vec2 +
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# Augmentation: SpecAugment
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# Authors: Le Do Thanh Binh 2021
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# ################################
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# ################################
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# Model: wav2vec2 + CTC
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# Augmentation: SpecAugment
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# Authors: Le Do Thanh Binh 2021
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# ################################
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model.ckpt
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