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import gradio as gr | |
import time | |
import whisper | |
# =============== User defined data ======================= | |
model_type = "base" | |
# =================== Model loading =================== | |
model = whisper.load_model(model_type) | |
# =================== Inference python =================== | |
def transcribe(audio): | |
# load audio and pad/trim it to fit 30 seconds | |
audio = whisper.load_audio(audio) | |
audio = whisper.pad_or_trim(audio) | |
# make log-Mel spectrogram and move to the same device as the model | |
mel = whisper.log_mel_spectrogram(audio).to(model.device) | |
# detect the spoken language | |
_, probs = model.detect_language(mel) | |
print(f"Detected language: {max(probs, key=probs.get)}") | |
# decode the audio | |
options = whisper.DecodingOptions(fp16 = False) | |
result = whisper.decode(model, mel, options) | |
return result.text | |
# =================== Run UI =================== | |
gr.Interface( | |
title = 'OpenAI Whisper ASR Gradio Web UI', | |
fn=transcribe, | |
inputs=[ | |
gr.inputs.Audio(source="microphone", type="filepath") | |
], | |
outputs=[ | |
"textbox" | |
], | |
live=True).launch(enable_queue=True) | |