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Browse files- app.py +14 -8
- requirements.txt +1 -2
app.py
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@@ -1,21 +1,24 @@
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import gradio as gr
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import spaces
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import torch
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import librosa
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from transformers import Wav2Vec2FeatureExtractor, Wav2Vec2ForSequenceClassification
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device = torch.device('cuda:0' if torch.cuda.is_available() else 'cpu')
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model_name = "Hemg/human-emotion-detection"
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feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(model_name)
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model = Wav2Vec2ForSequenceClassification.from_pretrained(model_name).to(device)
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def preprocess_audio(audio):
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audio_array, sampling_rate = librosa.load(audio, sr=16000) # Load and resample to 16kHz
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return {'speech': audio_array, 'sampling_rate': sampling_rate}
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@spaces.GPU
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def inference(audio):
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example = preprocess_audio(audio)
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inputs = feature_extractor(example['speech'], sampling_rate=16000, return_tensors="pt", padding=True)
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inputs = inputs.to(device) # Move inputs to GPU
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@@ -23,12 +26,15 @@ def inference(audio):
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logits = model(**inputs).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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return model.config.id2label[predicted_ids.item()], logits, predicted_ids # Move tensors back to CPU for further processing
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iface = gr.Interface(fn=
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inputs=gr.
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outputs="
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title="Audio Sentiment Analysis",
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description="Upload an audio file or record one to analyze sentiment.")
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iface.launch()
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import gradio as gr
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import spaces
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import torch
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#import librosa
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#import numpy as np
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from transformers import Wav2Vec2FeatureExtractor, Wav2Vec2ForSequenceClassification
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device = torch.device('cuda:0' if torch.cuda.is_available() else 'cpu')
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model_name = "Hemg/human-emotion-detection"
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feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(model_name)
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model = Wav2Vec2ForSequenceClassification.from_pretrained(model_name).to(device)
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def preprocess_audio(audio):
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#audio_array, sampling_rate = librosa.load(audio, sr=16000) # Load and resample to 16kHz
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#return {'speech': audio_array, 'sampling_rate': sampling_rate}
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@spaces.GPU
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def inference(audio):
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print('hello')
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'''
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example = preprocess_audio(audio)
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inputs = feature_extractor(example['speech'], sampling_rate=16000, return_tensors="pt", padding=True)
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inputs = inputs.to(device) # Move inputs to GPU
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logits = model(**inputs).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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return model.config.id2label[predicted_ids.item()], logits, predicted_ids # Move tensors back to CPU for further processing
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'''
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iface = gr.Interface(fn=inference,
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inputs=gr.Audio(type="filepath"),
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outputs=[gr.Label(label="Predicted Sentiment"),
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gr.JSON(label="Logits"),
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gr.JSON(label="Predicted ID")],
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title="Audio Sentiment Analysis",
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description="Upload an audio file or record one to analyze sentiment.")
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iface.launch(share=True)
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requirements.txt
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@@ -1,4 +1,3 @@
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gradio
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torch
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transformers
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torch
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transformers
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accelerate
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