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| import torch | |
| from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline | |
| import gradio as gr | |
| import torchaudio | |
| device = "cuda:0" if torch.cuda.is_available() else "cpu" | |
| torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 | |
| model_id = "distil-whisper/distil-large-v3" | |
| model = AutoModelForSpeechSeq2Seq.from_pretrained( | |
| model_id, torch_dtype=torch_dtype, use_safetensors=True | |
| ) | |
| model.to(device) | |
| processor = AutoProcessor.from_pretrained(model_id) | |
| pipe = pipeline( | |
| "automatic-speech-recognition", | |
| model=model, | |
| tokenizer=processor.tokenizer, | |
| feature_extractor=processor.feature_extractor, | |
| max_new_tokens=128, | |
| chunk_length_s=25, | |
| batch_size=16, | |
| torch_dtype=torch_dtype, | |
| device=device, | |
| ) | |
| def speech_to_text(audio_file): | |
| try: | |
| waveform, sample_rate = torchaudio.load(audio_file) | |
| if waveform.size(0) > 1: | |
| resample = torchaudio.transforms.Resample(sample_rate, sample_rate) | |
| waveform = resample(waveform) | |
| waveform_np = waveform.numpy() | |
| print("pass to pipe") | |
| result = pipe(waveform_np[0]) | |
| print("result",result) | |
| return result["text"] | |
| except Exception as e: | |
| print(f"Error: {str(e)}") | |
| iface = gr.Interface(fn=speech_to_text, inputs="file", outputs="text", title="Speech-to-Text") | |
| if __name__ == "__main__": | |
| iface.launch() |