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Update app.py
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app.py
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@@ -5,6 +5,7 @@ subprocess.run(["pip", "install", "torch", "torchvision", "torchaudio", "-f", "h
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import gradio as gr
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from transformers import WhisperProcessor, WhisperForConditionalGeneration
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# Load model and processor
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processor = WhisperProcessor.from_pretrained("openai/whisper-large")
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@@ -13,8 +14,16 @@ model.config.forced_decoder_ids = None
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# Function to perform ASR on audio data
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def transcribe_audio(audio_data):
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# Apply custom preprocessing to the audio data if needed
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processed_input = processor(audio_data, return_tensors="pt").input_features
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# Generate token ids
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predicted_ids = model.generate(processed_input)
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import gradio as gr
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from transformers import WhisperProcessor, WhisperForConditionalGeneration
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import torchaudio
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# Load model and processor
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processor = WhisperProcessor.from_pretrained("openai/whisper-large")
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# Function to perform ASR on audio data
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def transcribe_audio(audio_data):
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# Convert audio data to mono and normalize
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audio_data = torchaudio.functional.to_mono(audio_data)
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audio_data = torchaudio.functional.gain(audio_data, gain_db=5.0)
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# Resample if needed (Whisper model requires 16 kHz sampling rate)
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if audio_data[1] != 16000:
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audio_data = torchaudio.transforms.Resample(audio_data[1], 16000)(audio_data[0])
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# Apply custom preprocessing to the audio data if needed
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processed_input = processor(audio_data[0].numpy(), return_tensors="pt").input_features
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# Generate token ids
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predicted_ids = model.generate(processed_input)
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