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README.md
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pipeline_tag: automatic-speech-recognition
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tags:
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- CTC
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- Attention
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- pytorch
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- speechbrain
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- Transformer
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<iframe src="https://ghbtns.com/github-btn.html?user=speechbrain&repo=speechbrain&type=star&count=true&size=large&v=2" frameborder="0" scrolling="0" width="170" height="30" title="GitHub"></iframe>
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<br/><br/>
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# wav2vec 2.0 with CTC
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This repository provides all the necessary tools to perform automatic speech
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recognition from an end-to-end system pretrained on CommonVoice (English Language) within
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This ASR system is composed of 2 different but linked blocks:
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- Tokenizer (unigram) that transforms words into subword units and trained with
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the train transcriptions (train.tsv) of CommonVoice (EN).
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- Acoustic model (wav2vec2.0 + CTC
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The obtained final acoustic representation is given to the CTC
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The system is trained with recordings sampled at 16kHz (single channel).
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The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling *transcribe_file* if needed.
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pipeline_tag: automatic-speech-recognition
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tags:
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- CTC
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- pytorch
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- speechbrain
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- Transformer
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<iframe src="https://ghbtns.com/github-btn.html?user=speechbrain&repo=speechbrain&type=star&count=true&size=large&v=2" frameborder="0" scrolling="0" width="170" height="30" title="GitHub"></iframe>
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<br/><br/>
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# wav2vec 2.0 with CTC trained on CommonVoice English (No LM)
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This repository provides all the necessary tools to perform automatic speech
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recognition from an end-to-end system pretrained on CommonVoice (English Language) within
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This ASR system is composed of 2 different but linked blocks:
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- Tokenizer (unigram) that transforms words into subword units and trained with
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the train transcriptions (train.tsv) of CommonVoice (EN).
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- Acoustic model (wav2vec2.0 + CTC). A pretrained wav2vec 2.0 model ([wav2vec2-lv60-large](https://huggingface.co/facebook/wav2vec2-large-lv60)) is combined with two DNN layers and finetuned on CommonVoice En.
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The obtained final acoustic representation is given to the CTC decoder.
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The system is trained with recordings sampled at 16kHz (single channel).
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The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling *transcribe_file* if needed.
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