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Update app.py

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  1. app.py +10 -8
app.py CHANGED
@@ -26,24 +26,26 @@ ui.theme = "peach"
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  ui.article = """<h2>Pre-trained model Information</h2>
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  <h3>Automatic Speech Recognition</h3>
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  <p style='text-align: justify'>The model used for the ASR part of this space is from
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- [https://huggingface.co/facebook/hubert-large-ls960-ft] which is pretrained and fine-tuned on <b>960 hours of
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  Librispeech</b> on 16kHz sampled speech audio. This model has a self-reported <b>word error rate (WER)</b> of <b>1.9
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  percent</b> and ranks first in <i>paperswithcode</i> for ASR on Librispeech. More information can be
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- found on its website at [https://ai.facebook.com/blog/hubert-self-supervised-representation-learning-for-speech-recognition-
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- generation-and-compression] and
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- original model is under [https://github.com/pytorch/fairseq/tree/main/examples/hubert].</p>
 
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  <h3>Text Translator</h3>
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  <p style='text-align: justify'>The English to Spanish text translator pre-trained model is from
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- [https://huggingface.co/Helsinki-NLP/opus-mt-en-es] which is part of the <b>The Tatoeba Translation Challenge
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  (v2021-08-07)</b> as seen from its github repo at
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- [https://github.com/Helsinki-NLP/Tatoeba-Challenge]. This project aims to develop machine
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  translation in real-world
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  cases for many languages. </p>
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  <h3>Text to Speech</h3>
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- <p style='text-align: justify'> The TTS model used is from [https://huggingface.co/facebook/tts_transformer-es-css10].
 
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  This model uses the <b>Fairseq(-py)</b> sequence modeling toolkit for speech synthesis, in this case, specifically TTS
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  for Spanish. More information can be seen on their git at
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- [https://github.com/pytorch/fairseq/tree/main/examples/speech_synthesis]. </p>
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  """
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  ui.article = """<h2>Pre-trained model Information</h2>
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  <h3>Automatic Speech Recognition</h3>
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  <p style='text-align: justify'>The model used for the ASR part of this space is from
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+ <a href=\"https://huggingface.co/facebook/hubert-large-ls960-ft"> which is pretrained and fine-tuned on <b>960 hours of
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  Librispeech</b> on 16kHz sampled speech audio. This model has a self-reported <b>word error rate (WER)</b> of <b>1.9
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  percent</b> and ranks first in <i>paperswithcode</i> for ASR on Librispeech. More information can be
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+ found on its website at <a href=\"https://ai.facebook.com/blog/hubert-self-supervised-representation-learning-for-speech-
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+ recognition-
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+ generation-and-compression"> and
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+ original model is under <a href=\"https://github.com/pytorch/fairseq/tree/main/examples/hubert">.</p>
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  <h3>Text Translator</h3>
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  <p style='text-align: justify'>The English to Spanish text translator pre-trained model is from
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+ <a href=\"https://huggingface.co/Helsinki-NLP/opus-mt-en-es"> which is part of the <b>The Tatoeba Translation Challenge
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  (v2021-08-07)</b> as seen from its github repo at
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+ <a href=\"https://github.com/Helsinki-NLP/Tatoeba-Challenge">. This project aims to develop machine
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  translation in real-world
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  cases for many languages. </p>
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  <h3>Text to Speech</h3>
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+ <p style='text-align: justify'> The TTS model used is from <a href=\"https://huggingface.co/facebook/tts_transformer-es-
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+ css10">.
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  This model uses the <b>Fairseq(-py)</b> sequence modeling toolkit for speech synthesis, in this case, specifically TTS
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  for Spanish. More information can be seen on their git at
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+ <a href=\"https://github.com/pytorch/fairseq/tree/main/examples/speech_synthesis">. </p>
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  """
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