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Update app.py
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app.py
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import os
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from transformers import
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import gradio as gr
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import numpy as np
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# Fetch the token from the environment
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hf_token = os.getenv("HUGGINGFACE_HUB_TOKEN")
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# Convert audio to floating-point if necessary
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if audio_input.dtype != np.float32:
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audio_input = audio_input.astype(np.float32)
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# Resample if the sample rate is not 16000
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if sr != 16000:
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audio_input = librosa.resample(audio_input, orig_sr=sr, target_sr=16000)
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# Process and transcribe the audio
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input_features = processor(audio_input, sampling_rate=16000, return_tensors="pt").input_features
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# Generate predictions
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predicted_ids = model.generate(input_features)
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transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)
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text = transcription
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return text
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# Create the Gradio interface
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iface = gr.Interface(
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fn=transcribe,
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inputs=gr.Audio(),
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outputs="text",
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title="Whisper Medium Shqip",
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description="Realtime demo for Sq speech recognition using a fine-tuned Whisper medium model.",
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import os
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from transformers import pipeline
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import gradio as gr
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# Fetch the token from the environment
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hf_token = os.getenv("HUGGINGFACE_HUB_TOKEN")
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model_id = "akadriu/whisper-medium-sq" # update with your model id
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pipe = pipeline("automatic-speech-recognition", model=model_id, token=hf_token)
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def transcribe_speech(filepath):
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output = pipe(
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filepath,
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max_new_tokens=256,
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generate_kwargs={
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"task": "transcribe",
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"language": "albanian",
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}, # update with the language you've fine-tuned on
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chunk_length_s=30,
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batch_size=8,
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)
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return output["text"]
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import gradio as gr
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demo = gr.Blocks()
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mic_transcribe = gr.Interface(
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fn=transcribe_speech,
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inputs=gr.Audio(sources="microphone", type="filepath"),
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outputs=gr.outputs.Textbox(),
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)
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file_transcribe = gr.Interface(
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fn=transcribe_speech,
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inputs=gr.Audio(sources="upload", type="filepath"),
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outputs=gr.outputs.Textbox(),
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)
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with demo:
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gr.TabbedInterface(
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[mic_transcribe, file_transcribe],
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["Transcribe Microphone", "Transcribe Audio File"],
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)
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demo.launch(debug=True)
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