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Update app.py
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app.py
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@@ -4,7 +4,7 @@ import torch
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from datasets import load_dataset
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from transformers import SpeechT5ForTextToSpeech, SpeechT5HifiGan, SpeechT5Processor, pipeline
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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@@ -12,10 +12,12 @@ device = "cuda:0" if torch.cuda.is_available() else "cpu"
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asr_pipe = pipeline("automatic-speech-recognition", model="openai/whisper-base", device=device)
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# load text-to-speech checkpoint and speaker embeddings
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processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts")
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model = SpeechT5ForTextToSpeech.from_pretrained("sanchit-gandhi/speecht5_tts_vox_nl").to(device)
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vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device)
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embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation")
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speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0)
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@@ -25,11 +27,18 @@ def translate(audio):
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outputs = asr_pipe(audio, max_new_tokens=256, generate_kwargs={"task": "transcribe", "language": "nl"})
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return outputs["text"]
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def synthesise(text):
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inputs =
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return speech.cpu()
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def speech_to_speech_translation(audio):
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from datasets import load_dataset
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from transformers import SpeechT5ForTextToSpeech, SpeechT5HifiGan, SpeechT5Processor, pipeline
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from transformers import VitsModel, VitsTokenizer
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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asr_pipe = pipeline("automatic-speech-recognition", model="openai/whisper-base", device=device)
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# load text-to-speech checkpoint and speaker embeddings
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# processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts")
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# model = SpeechT5ForTextToSpeech.from_pretrained("sanchit-gandhi/speecht5_tts_vox_nl").to(device)
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vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device)
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model = VitsModel.from_pretrained("Matthijs/mms-tts-nld")
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tokenizer = VitsTokenizer.from_pretrained("Matthijs/mms-tts-nld")
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embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation")
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speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0)
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outputs = asr_pipe(audio, max_new_tokens=256, generate_kwargs={"task": "transcribe", "language": "nl"})
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return outputs["text"]
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def synthesise(text):
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inputs = tokenizer(text, return_tensors="pt")
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with torch.no_grad():
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outputs = model(inputs["input_ids"])
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speech = outputs.audio[0]
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return speech.cpu()
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# def synthesise(text):
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# inputs = processor(text=text, return_tensors="pt", padding='max_length', truncation=True)
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# speech = model.generate_speech(inputs["input_ids"].to(device), speaker_embeddings.to(device), vocoder=vocoder)
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# return speech.cpu()
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def speech_to_speech_translation(audio):
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