Update app.py
Browse files
app.py
CHANGED
@@ -22,87 +22,6 @@ model = VitsModel.from_pretrained("Matthijs/mms-tts-deu")
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tokenizer = VitsTokenizer.from_pretrained("Matthijs/mms-tts-deu")
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def translate(audio):
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outputs = asr_pipe(
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audio,
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max_new_tokens=256,
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generate_kwargs={"task": "transcribe", "language": "de"}
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)
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return outputs["text"]
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def synthesise(text):
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if len(text.strip()) == 0:
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return (16000, np.zeros(0).astype(np.int16))
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inputs = tokenizer(text, return_tensors="pt")
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input_ids = inputs["input_ids"]
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with torch.no_grad():
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outputs = model(input_ids)
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speech = outputs.audio[0]
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return speech.cpu()
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def speech_to_speech_translation(audio):
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translated_text = translate(audio)
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synthesised_speech = synthesise(translated_text)
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synthesised_speech = (synthesised_speech.numpy() * 32767).astype(np.int16)
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return 16000, synthesised_speech
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title = "Cascaded STST"
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description = """
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Demo for cascaded speech-to-speech translation (STST), mapping from source speech in any language to target speech in English. Demo uses OpenAI's [Whisper Base](https://huggingface.co/openai/whisper-base) model for speech translation, and Microsoft's
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[SpeechT5 TTS](https://huggingface.co/microsoft/speecht5_tts) model for text-to-speech:
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![Cascaded STST](https://huggingface.co/datasets/huggingface-course/audio-course-images/resolve/main/s2st_cascaded.png "Diagram of cascaded speech to speech translation")
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"""
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demo = gr.Blocks()
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mic_translate = gr.Interface(
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fn=speech_to_speech_translation,
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inputs=gr.Audio(source="microphone", type="filepath"),
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outputs=gr.Audio(label="Generated Speech", type="numpy"),
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title=title,
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description=description,
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)
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file_translate = gr.Interface(
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fn=speech_to_speech_translation,
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inputs=gr.Audio(source="upload", type="filepath"),
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outputs=gr.Audio(label="Generated Speech", type="numpy"),
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examples=[["./example.wav"]],
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title=title,
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description=description,
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)
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with demo:
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gr.TabbedInterface([mic_translate, file_translate], ["Microphone", "Audio File"])
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demo.launch()import gradio as gr
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import numpy as np
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import torch
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from transformers import (
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VitsModel,
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VitsTokenizer,
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pipeline
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)
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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# load speech translation checkpoint
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asr_pipe = pipeline(
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"automatic-speech-recognition",
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model="openai/whisper-base",
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device=device
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)
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model = VitsModel.from_pretrained("Matthijs/mms-tts-deu")
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tokenizer = VitsTokenizer.from_pretrained("Matthijs/mms-tts-deu")
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def translate(audio):
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outputs = asr_pipe(
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audio,
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tokenizer = VitsTokenizer.from_pretrained("Matthijs/mms-tts-deu")
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def translate(audio):
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outputs = asr_pipe(
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audio,
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