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Browse files- app.py +26 -9
- artst/__pycache__/__init__.cpython-38.pyc +0 -0
- artst/criterions/__pycache__/__init__.cpython-38.pyc +0 -0
- artst/criterions/__pycache__/artst_criterion.cpython-38.pyc +0 -0
- artst/criterions/__pycache__/speech_pretrain_criterion.cpython-38.pyc +0 -0
- artst/criterions/__pycache__/speech_to_text_loss.cpython-38.pyc +0 -0
- artst/criterions/__pycache__/text_pretrain_criterion.cpython-38.pyc +0 -0
- artst/criterions/__pycache__/text_to_speech_loss.cpython-38.pyc +0 -0
- artst/data/__pycache__/__init__.cpython-38.pyc +0 -0
- artst/data/__pycache__/multitask_dataset.cpython-38.pyc +0 -0
- artst/data/__pycache__/speech_dataset.cpython-38.pyc +0 -0
- artst/data/__pycache__/speech_to_class_dataset.cpython-38.pyc +0 -0
- artst/data/__pycache__/speech_to_speech_dataset.cpython-38.pyc +0 -0
- artst/data/__pycache__/speech_to_text_dataset.cpython-38.pyc +0 -0
- artst/data/__pycache__/text_dataset.cpython-38.pyc +0 -0
- artst/data/__pycache__/text_to_speech_dataset.cpython-38.pyc +0 -0
- artst/models/__pycache__/__init__.cpython-38.pyc +0 -0
- artst/models/__pycache__/artst.cpython-38.pyc +0 -0
- artst/models/__pycache__/t5_transformer_lm.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/__init__.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/decoder.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/encoder.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/multihead_attention.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/speaker_decoder_postnet.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/speech_decoder_postnet.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/speech_decoder_prenet.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/speech_encoder_postnet.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/speech_encoder_prenet.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/text_decoder_postnet.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/text_decoder_prenet.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/text_encoder_prenet.cpython-38.pyc +0 -0
- artst/models/modules/__pycache__/transformer_layer.cpython-38.pyc +0 -0
- artst/tasks/__pycache__/__init__.cpython-38.pyc +0 -0
- artst/tasks/__pycache__/artst.cpython-38.pyc +0 -0
- requirements.txt → pre-requirements.txt +0 -0
app.py
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@@ -12,15 +12,12 @@ from fairseq.tasks.hubert_pretraining import LabelEncoder
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from fairseq.data.audio.speech_to_text_dataset import get_features_or_waveform
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device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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WORK_DIR = os.getcwd()
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checkpoint = torch.load('ckpts/clartts_tts.pt')
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checkpoint['cfg']['task'].t5_task = 't2s'
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checkpoint['cfg']['task'].hubert_label_dir = "utils/"
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checkpoint['cfg']['task'].bpe_tokenizer = "utils/arabic.model"
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checkpoint['cfg']['task'].data = "utils/"
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checkpoint['cfg']['model'].mask_prob = 0.
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checkpoint['cfg']['task'].mask_prob = 0.
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task = ArTSTTask.setup_task(checkpoint['cfg']['task'])
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emb_path='embs/clartts.npy'
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@@ -56,20 +53,40 @@ def inference(text, spkr=emb_path):
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)
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with torch.no_grad():
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gen_audio = vocoder(outs.to(device))
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text_box = gr.Textbox(max_lines=2, label="Arabic Text", rtl=True)
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out = gr.Audio(label="Synthesized Audio", type="numpy")
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title="ArTST: Arabic Speech Synthesis"
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description="ArTST: Arabic text and speech transformer based on the T5 transformer. This space demonstarates the TTS checkpoint finetuned on \
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the CLARTTS dataset. The model is pre-trained on the MGB-2 dataset.
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examples=["لأن فراق المألوف في العادة ومجانبة ما صار متفقا عليه بالمواضعة",\
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"ومن لطيف حكمته أن جعل لكل عبادة حالتين",\
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"فمن لهم عدل الإنسان مع من فوقه"]
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demo = gr.Interface(inference, \
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inputs=text_box, outputs=out, title=title, description=description, examples=examples)
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if __name__ == "__main__":
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demo.launch(share=True)
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from fairseq.data.audio.speech_to_text_dataset import get_features_or_waveform
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device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
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checkpoint = torch.load('ckpts/clartts_tts.pt')
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checkpoint['cfg']['task'].t5_task = 't2s'
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checkpoint['cfg']['task'].bpe_tokenizer = "utils/arabic.model"
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checkpoint['cfg']['task'].data = "utils/"
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checkpoint['cfg']['model'].mask_prob = 0.5
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checkpoint['cfg']['task'].mask_prob = 0.5
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task = ArTSTTask.setup_task(checkpoint['cfg']['task'])
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emb_path='embs/clartts.npy'
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)
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with torch.no_grad():
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gen_audio = vocoder(outs.to(device))
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speech = (gen_audio.cpu().numpy() * 32767).astype(np.int16)
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return (16000,speech)
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text_box = gr.Textbox(max_lines=2, label="Arabic Text", rtl=True)
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out = gr.Audio(label="Synthesized Audio", type="numpy")
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title="ArTST: Arabic Speech Synthesis"
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description="ArTST: Arabic text and speech transformer based on the T5 transformer. This space demonstarates the TTS checkpoint finetuned on \
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the Classical Arabic Text-To-Speech (CLARTTS) dataset. The model is pre-trained on the MGB-2 dataset."
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examples=["لأن فراق المألوف في العادة ومجانبة ما صار متفقا عليه بالمواضعة",\
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"ومن لطيف حكمته أن جعل لكل عبادة حالتين",\
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"فمن لهم عدل الإنسان مع من فوقه"]
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article = """
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<div style='margin:20px auto;'>
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<p>References: <a href="https://arxiv.org/abs/2310.16621">ArTST paper</a> |
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<a href="https://github.com/mbzuai-nlp/ArTST">GitHub</a> |
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<a href="https://huggingface.co/MBZUAI/ArTST">Weights and Tokenizer</a></p>
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<pre>
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@misc{toyin2023artst,
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title={ArTST: Arabic Text and Speech Transformer},
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author={Hawau Olamide Toyin and Amirbek Djanibekov and Ajinkya Kulkarni and Hanan Aldarmaki},
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year={2023},
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eprint={2310.16621},
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archivePrefix={arXiv},
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primaryClass={cs.CL}
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}
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</pre>
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<p>Speaker embeddings were generated from <a href="http://www.festvox.org/cmu_arctic/">CMU ARCTIC</a>.</p>
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</div>
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"""
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demo = gr.Interface(inference, \
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inputs=text_box, outputs=out, title=title, description=description, examples=examples, article=article)
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if __name__ == "__main__":
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demo.launch(share=True)
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requirements.txt → pre-requirements.txt
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