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Create app.py
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app.py
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import gradio as gr
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import torch
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import numpy as np
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from transformers import pipeline, WhisperForConditionalGeneration, WhisperProcessor
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from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan
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from datasets import load_dataset
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# Check if a GPU is available and set the device
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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# Load the Whisper ASR model
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whisper_model_id = "riteshkr/quantized-whisper-large-v3"
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whisper_model = WhisperForConditionalGeneration.from_pretrained(whisper_model_id)
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whisper_processor = WhisperProcessor.from_pretrained(whisper_model_id)
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# Set the language to English using forced_decoder_ids
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forced_decoder_ids = whisper_processor.get_decoder_prompt_ids(language="english", task="transcribe")
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whisper_pipe = pipeline(
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"automatic-speech-recognition",
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model=whisper_model,
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tokenizer=whisper_processor.tokenizer,
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feature_extractor=whisper_processor.feature_extractor,
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device=0 if torch.cuda.is_available() else -1
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)
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# Load the SpeechT5 TTS model
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tts_processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts")
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tts_model = SpeechT5ForTextToSpeech.from_pretrained("microsoft/speecht5_tts")
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vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan")
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tts_model.to(device)
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vocoder.to(device)
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# Load speaker embeddings for TTS
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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).to(device)
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# Set target data type and max range for speech
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target_dtype = np.int16
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max_range = np.iinfo(target_dtype).max
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# Define the transcription function (Whisper ASR)
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def transcribe_speech(filepath):
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batch_size = 16 if torch.cuda.is_available() else 4
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output = whisper_pipe(
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filepath,
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max_new_tokens=256,
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generate_kwargs={"forced_decoder_ids": forced_decoder_ids},
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chunk_length_s=30,
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batch_size=batch_size,
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)
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return output["text"]
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# Define the synthesis function (SpeechT5 TTS)
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def synthesise(text):
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inputs = tts_processor(text=text, return_tensors="pt")
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speech = tts_model.generate_speech(
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inputs["input_ids"].to(device), speaker_embeddings, vocoder=vocoder
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)
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return speech.cpu()
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# Define the speech-to-speech translation function
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def speech_to_speech_translation(audio):
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# Transcribe speech
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translated_text = transcribe_speech(audio)
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# Synthesize speech
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synthesised_speech = synthesise(translated_text)
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# Convert speech to desired format
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synthesised_speech = (synthesised_speech.numpy() * max_range).astype(np.int16)
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return 16000, synthesised_speech
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# Define the Gradio interfaces for microphone input and file upload
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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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)
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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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)
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# Define the Gradio interfaces for transcription
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mic_transcribe = gr.Interface(
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fn=transcribe_speech,
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inputs=gr.Audio(source="microphone", type="filepath"),
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outputs=gr.Textbox(),
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)
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file_transcribe = gr.Interface(
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fn=transcribe_speech,
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inputs=gr.Audio(source="upload", type="filepath"),
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outputs=gr.Textbox(),
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)
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# Create the app using Gradio Blocks with tabbed interfaces
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demo = gr.Blocks()
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with demo:
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gr.TabbedInterface(
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[
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mic_transcribe, file_transcribe, # For transcription
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mic_translate, file_translate # For speech-to-speech translation
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],
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[
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"Transcribe Microphone", "Transcribe Audio File",
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"Translate Microphone", "Translate Audio File"
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]
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)
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# Launch the app with debugging enabled
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if __name__ == "__main__":
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demo.launch(debug=True, share=True)
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