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| import subprocess | |
| subprocess.run(["python", "-m", "pip", "install", "--upgrade", "pip"]) | |
| subprocess.run(["pip", "install", "gradio", "--upgrade"]) | |
| subprocess.run(["pip", "install", "datasets"]) | |
| subprocess.run(["pip", "install", "transformers"]) | |
| subprocess.run(["pip", "install", "torch", "torchvision", "torchaudio", "-f", "https://download.pytorch.org/whl/torch_stable.html"]) | |
| import gradio as gr | |
| import numpy as np | |
| from transformers import WhisperProcessor, WhisperForConditionalGeneration | |
| # Load model and processor | |
| processor = WhisperProcessor.from_pretrained("openai/whisper-large") | |
| model = WhisperForConditionalGeneration.from_pretrained("openai/whisper-large") | |
| forced_decoder_ids = processor.get_decoder_prompt_ids(language="italian", task="transcribe") | |
| # Custom preprocessing function | |
| def preprocess_audio(audio_data, sampling_rate=16_000): | |
| print(type(audio_data)) | |
| print(audio_data) | |
| raw_speech = np.asarray(audio_data, dtype=np.float32) | |
| # Pad or truncate the audio data to the required length | |
| if len(raw_speech) > processor.feature_extractor.max_len: | |
| raw_speech = raw_speech[:processor.feature_extractor.max_len] | |
| else: | |
| raw_speech = np.pad(raw_speech, (0, processor.feature_extractor.max_len - len(raw_speech))) | |
| # Process the audio data using the Whisper processor | |
| processed_data = processor( | |
| raw_speech, | |
| sampling_rate=sampling_rate, | |
| return_tensors="pt", | |
| padding=True, | |
| truncation=True | |
| ) | |
| return processed_data.input_features | |
| # Function to perform ASR on audio data | |
| def transcribe_audio(audio_data): | |
| # Preprocess the audio data | |
| input_features = preprocess_audio(audio_data) | |
| # Generate token ids | |
| predicted_ids = model.generate(input_features) | |
| # Decode token ids to text | |
| transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True) | |
| return transcription[0] | |
| # Create Gradio interface | |
| audio_input = gr.Audio() | |
| gr.Interface(fn=transcribe_audio, inputs=audio_input, outputs="text").launch() | |