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| import torch | |
| from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline | |
| import gradio as gr | |
| 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-small.en" | |
| 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, | |
| torch_dtype=torch_dtype, | |
| device=device, | |
| ) | |
| def audio2text(audio_file): | |
| output=pipe(audio_file) | |
| return output['text'] | |
| gr.Interface(fn=audio2text, inputs=[gr.Audio(label='upload your audio file', sources='upload', type='filepath')], outputs=[gr.Textbox(label="transcription")]).launch() | |