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Update app.py
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app.py
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@@ -1,6 +1,7 @@
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import gradio as gr
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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import torch
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# Load the pre-trained model and processor
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model_name = "facebook/s2t-wav2vec2-large-en-ar"
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@@ -9,13 +10,20 @@ processor = Wav2Vec2Processor.from_pretrained(model_name)
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# Define a function for the ASR model
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def transcribe(audio):
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# Process the audio
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inputs = processor(audio, return_tensors="pt", sampling_rate=16000)
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# Get the model's predictions
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logits = model(input_values=inputs.input_values).logits
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# Decode the predicted text
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predicted_ids = logits.argmax(dim=-1)
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transcription = processor.decode(predicted_ids[0])
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return transcription
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# Define the Gradio interface
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import gradio as gr
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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import torch
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import numpy as np
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# Load the pre-trained model and processor
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model_name = "facebook/s2t-wav2vec2-large-en-ar"
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# Define a function for the ASR model
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def transcribe(audio):
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# Convert the audio into a format compatible with the processor
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if isinstance(audio, np.ndarray):
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audio = audio.flatten() # Ensure it's a 1D array
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# Process the audio
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inputs = processor(audio, return_tensors="pt", sampling_rate=16000)
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# Get the model's predictions
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logits = model(input_values=inputs.input_values).logits
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# Decode the predicted text
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predicted_ids = logits.argmax(dim=-1)
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transcription = processor.decode(predicted_ids[0])
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return transcription
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# Define the Gradio interface
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