nguyenvulebinh
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add sample audio file
Browse files- .ipynb_checkpoints/README-checkpoint.md +83 -0
- README.md +81 -1
.ipynb_checkpoints/README-checkpoint.md
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---
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language: vi
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datasets:
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- common_voice
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- librispeech_asr
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- how2
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- must-c-v1
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- must-c-v2
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- europarl
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- tedlium
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tags:
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- audio
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- automatic-speech-recognition
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license: cc-by-nc-4.0
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---
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# Fine-Tune Wav2Vec2 large model for English ASR
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### Data for fine-tune
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| Dataset | Duration in hours |
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|--------------|-------------------|
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| Common Voice | 1667 |
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| Europarl | 85 |
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| How2 | 356 |
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| Librispeech | 936 |
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| MuST-C v1 | 407 |
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| MuST-C v2 | 482 |
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| Tedlium | 482 |
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### Evaluation result
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| Dataset | Duration in hours | WER w/o LM | WER with LM |
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|-------------|-------------------|------------|-------------|
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| Librispeech | 5.4 | 2.9 | 1.1 |
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| Tedlium | 2.6 | 7.9 | 5.4 |
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### Usage
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[![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/drive/1FAhtGvjRdHT4W0KeMdMMlL7sm6Hbe7dv?usp=sharing)
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```python
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from transformers.file_utils import cached_path, hf_bucket_url
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from importlib.machinery import SourceFileLoader
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from transformers import Wav2Vec2ProcessorWithLM
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from IPython.lib.display import Audio
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import torchaudio
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import torch
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# Load model & processor
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model_name = "nguyenvulebinh/iwslt-asr-wav2vec-large"
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model = SourceFileLoader("model", cached_path(hf_bucket_url(model_name,filename="model_handling.py"))).load_module().Wav2Vec2ForCTC.from_pretrained(model_name)
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processor = Wav2Vec2ProcessorWithLM.from_pretrained(model_name)
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# Load an example audio (16k)
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audio, sample_rate = torchaudio.load(cached_path(hf_bucket_url(model_name, filename="tst_2010_sample.wav")))
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input_data = processor.feature_extractor(audio[0], sampling_rate=16000, return_tensors='pt')
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# Infer
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output = model(**input_data)
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# Output transcript without LM
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print(processor.tokenizer.decode(output.logits.argmax(dim=-1)[0].detach().cpu().numpy()))
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# and of course there's teams that have a lot more tada structures and among the best are recent graduates of kindergarten
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# Output transcript with LM
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print(processor.decode(output.logits.cpu().detach().numpy()[0], beam_width=100).text)
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# and of course there are teams that have a lot more ta da structures and among the best are recent graduates of kindergarten
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```
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### Model Parameters License
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The ASR model parameters are made available for non-commercial use only, under the terms of the Creative Commons Attribution-NonCommercial 4.0 International (CC BY-NC 4.0) license. You can find details at: https://creativecommons.org/licenses/by-nc/4.0/legalcode
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### Contact
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nguyenvulebinh@gmail.com
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[![Follow](https://img.shields.io/twitter/follow/nguyenvulebinh?style=social)](https://twitter.com/intent/follow?screen_name=nguyenvulebinh)
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README.md
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1 |
---
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language: vi
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+
datasets:
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+
- common_voice
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+
- librispeech_asr
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+
- how2
|
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+
- must-c-v1
|
8 |
+
- must-c-v2
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- europarl
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- tedlium
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tags:
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- audio
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- automatic-speech-recognition
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license: cc-by-nc-4.0
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---
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+
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+
# Fine-Tune Wav2Vec2 large model for English ASR
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18 |
+
|
19 |
+
|
20 |
+
### Data for fine-tune
|
21 |
+
|
22 |
+
| Dataset | Duration in hours |
|
23 |
+
|--------------|-------------------|
|
24 |
+
| Common Voice | 1667 |
|
25 |
+
| Europarl | 85 |
|
26 |
+
| How2 | 356 |
|
27 |
+
| Librispeech | 936 |
|
28 |
+
| MuST-C v1 | 407 |
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29 |
+
| MuST-C v2 | 482 |
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30 |
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| Tedlium | 482 |
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31 |
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32 |
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### Evaluation result
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35 |
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| Dataset | Duration in hours | WER w/o LM | WER with LM |
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36 |
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|-------------|-------------------|------------|-------------|
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| Librispeech | 5.4 | 2.9 | 1.1 |
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| Tedlium | 2.6 | 7.9 | 5.4 |
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39 |
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### Usage
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42 |
+
|
43 |
+
[![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/drive/1FAhtGvjRdHT4W0KeMdMMlL7sm6Hbe7dv?usp=sharing)
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```python
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from transformers.file_utils import cached_path, hf_bucket_url
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47 |
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from importlib.machinery import SourceFileLoader
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from transformers import Wav2Vec2ProcessorWithLM
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from IPython.lib.display import Audio
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50 |
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import torchaudio
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import torch
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# Load model & processor
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model_name = "nguyenvulebinh/iwslt-asr-wav2vec-large"
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model = SourceFileLoader("model", cached_path(hf_bucket_url(model_name,filename="model_handling.py"))).load_module().Wav2Vec2ForCTC.from_pretrained(model_name)
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processor = Wav2Vec2ProcessorWithLM.from_pretrained(model_name)
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+
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# Load an example audio (16k)
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59 |
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audio, sample_rate = torchaudio.load(cached_path(hf_bucket_url(model_name, filename="tst_2010_sample.wav")))
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input_data = processor.feature_extractor(audio[0], sampling_rate=16000, return_tensors='pt')
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# Infer
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output = model(**input_data)
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# Output transcript without LM
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print(processor.tokenizer.decode(output.logits.argmax(dim=-1)[0].detach().cpu().numpy()))
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# and of course there's teams that have a lot more tada structures and among the best are recent graduates of kindergarten
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# Output transcript with LM
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print(processor.decode(output.logits.cpu().detach().numpy()[0], beam_width=100).text)
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# and of course there are teams that have a lot more ta da structures and among the best are recent graduates of kindergarten
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```
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### Model Parameters License
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75 |
+
|
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The ASR model parameters are made available for non-commercial use only, under the terms of the Creative Commons Attribution-NonCommercial 4.0 International (CC BY-NC 4.0) license. You can find details at: https://creativecommons.org/licenses/by-nc/4.0/legalcode
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### Contact
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+
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+
nguyenvulebinh@gmail.com
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+
[![Follow](https://img.shields.io/twitter/follow/nguyenvulebinh?style=social)](https://twitter.com/intent/follow?screen_name=nguyenvulebinh)
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