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# app.py | |
# ============= | |
# This is a complete app.py file for an automatic speech recognition app using the openai/whisper-large-v3-turbo model. | |
# The app is built using Gradio and Hugging Face Transformers, and it runs on the CPU to avoid video memory usage. | |
import torch | |
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline | |
import gradio as gr | |
# Set device to CPU | |
device = "cpu" | |
torch_dtype = torch.float32 | |
# Load the model and processor | |
model_id = "openai/whisper-large-v3-turbo" | |
model = AutoModelForSpeechSeq2Seq.from_pretrained( | |
model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True | |
) | |
model.to(device) | |
processor = AutoProcessor.from_pretrained(model_id) | |
# Create the ASR pipeline | |
pipe = pipeline( | |
"automatic-speech-recognition", | |
model=model, | |
tokenizer=processor.tokenizer, | |
feature_extractor=processor.feature_extractor, | |
torch_dtype=torch_dtype, | |
device=device, | |
) | |
def transcribe_audio(audio_file): | |
""" | |
Transcribe the given audio file using the Whisper model. | |
Parameters: | |
audio_file (str): Path to the audio file. | |
Returns: | |
str: Transcribed text. | |
""" | |
result = pipe(audio_file) | |
return result["text"] | |
# Define the Gradio interface | |
iface = gr.Interface( | |
fn=transcribe_audio, | |
inputs=gr.inputs.Audio(source="upload", type="filepath"), | |
outputs="text", | |
title="Whisper ASR Demo", | |
description="Upload an audio file and get the transcribed text using the openai/whisper-large-v3-turbo model.", | |
) | |
# Launch the Gradio app | |
if __name__ == "__main__": | |
iface.launch() |