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datasciencedojo
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13b9ce8
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Parent(s):
c572992
Create app.py
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app.py
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import torch
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import gradio as gr
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import pytube as pt
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from transformers import pipeline
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from huggingface_hub import model_info
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MODEL_NAME = "openai/whisper-small" #this always needs to stay in line 8 :D sorry for the hackiness
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lang = "en"
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device = 0 if torch.cuda.is_available() else "cpu"
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pipe = pipeline(
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task="automatic-speech-recognition",
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model=MODEL_NAME,
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chunk_length_s=30,
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device=device,
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)
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pipe.model.config.forced_decoder_ids = pipe.tokenizer.get_decoder_prompt_ids(language=lang, task="transcribe")
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def transcribe(microphone, file_upload):
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warn_output = ""
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if (microphone is not None) and (file_upload is not None):
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warn_output = (
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"WARNING: You've uploaded an audio file and used the microphone. "
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"The recorded file from the microphone will be used and the uploaded audio will be discarded.\n"
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)
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elif (microphone is None) and (file_upload is None):
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return "ERROR: You have to either use the microphone or upload an audio file"
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file = microphone if microphone is not None else file_upload
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text = pipe(file)["text"]
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return warn_output + text
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demo = gr.Blocks()
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examples = [
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['TestAudio1.mp3'], ['TestAudio2.wav'], ['TestAudio3.wav'], ['TestAudio4.wav'], ['TestAudio5.wav'], ['TestAudio6.wav'], ['TestAudio7.wav'], ['TestAudio8.wav'], ['TestAudio9.wav'], ['TestAudio10.wav']
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]
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mf_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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gr.inputs.Audio(source="microphone", type="filepath", optional=True),
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gr.inputs.Audio(source="upload", type="filepath", optional=True)
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],
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outputs="text",
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layout="horizontal",
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theme="huggingface",
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allow_flagging="never",
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examples = examples
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).launch(enable_queue=True)
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#used openai/whisper model
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