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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 numpy as np
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import torch
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from transformers import pipeline, AutoModelForSpeechSeq2Seq, AutoProcessor
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model_id = 'openai/whisper-large-v3'
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
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model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id, torch_dtype=torch_dtype).to(device)
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processor = AutoProcessor.from_pretrained(model_id)
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pipe_asr = pipeline("automatic-speech-recognition", model=model, tokenizer=processor.tokenizer, feature_extractor=processor.feature_extractor, max_new_tokens=128, chunk_length_s=15, batch_size=16, torch_dtype=torch_dtype, device=device, return_timestamps=True)
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base_audio_drive = "/data/audio"
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def transcribe_function(audio):
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sr, y = audio[0], audio[1]
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y = y.astype(np.float32) / np.max(np.abs(y))
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result = pipe_asr({"array": y, "sampling_rate": sr}, return_timestamps=False)
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full_text = result.get("text", "")
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return full_text
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def gradio_transcribe(audio):
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sr, y = audio
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return transcribe_function((sr, y))
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iface = gr.Interface(fn=gradio_transcribe, inputs=gr.inputs.Audio(source="microphone", type="numpy"), outputs="text", title="Voice to Text Transcription", description="Transcribe your voice input to text using a pre-trained Whisper model.")
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iface.launch()
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