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import gradio as gr
import numpy as np
import matplotlib.pyplot as plt
from PIL import Image
import librosa
import time
from datetime import datetime
import pandas as pd
HOME_DIR = ""
local_config_path = 'config.json'
local_preprocessor_config_path = 'preprocessor_config.json'
local_weights_path = 'pytorch_model.bin'
local_training_args_path = 'training_args.bin'
import torch
import torch.nn.functional as F
import numpy as np
from tqdm import tqdm
# Define the id2label mapping
id2label = {
0: "angry",
1: "disgust",
2: "fear",
3: "happy",
4: "neutral",
5: "sad",
6: "surprise"
}
def predict(model, feature_extractor, data, max_length, id2label):
# Extract features
print(datetime.now().strftime('%Y-%m-%d%H:%M:%S'), ":Extracting features...")
inputs = feature_extractor(data, sampling_rate=16000, max_length=max_length, return_tensors='tf', padding=True, truncation=True)
torch_inputs = torch.tensor(inputs['input_values'].numpy(), dtype=torch.float32)
print(datetime.now().strftime('%Y-%m-%d%H:%M:%S'), ":Predicting...")
# Forward pass
outputs = model(input_values=torch_inputs)
# Extract logits from the output
logits = outputs
# Apply softmax to get probabilities
probabilities = F.softmax(logits, dim=-1)
# Get the predicted class index
predicted_class_idx = torch.argmax(probabilities, dim=-1).item()
predicted_label = id2label[predicted_class_idx]
#predicted_label = predicted_class_idx
return predicted_label, probabilities
from transformers import Wav2Vec2Config, Wav2Vec2Model
import torch.nn as nn
from huggingface_hub import PyTorchModelHubMixin
config = Wav2Vec2Config.from_pretrained(local_config_path)
class Wav2Vec2ForSpeechClassification(nn.Module, PyTorchModelHubMixin):
def __init__(self, config):
super(Wav2Vec2ForSpeechClassification, self).__init__()
self.wav2vec2 = Wav2Vec2Model(config)
self.classifier = nn.ModuleDict({
'dense': nn.Linear(config.hidden_size, config.hidden_size),
'activation': nn.ReLU(),
'dropout': nn.Dropout(config.final_dropout),
'out_proj': nn.Linear(config.hidden_size, config.num_labels)
})
def forward(self, input_values):
outputs = self.wav2vec2(input_values)
hidden_states = outputs.last_hidden_state
x = self.classifier['dense'](hidden_states[:, 0, :])
x = self.classifier['activation'](x)
x = self.classifier['dropout'](x)
logits = self.classifier['out_proj'](x)
return logits
import json
from transformers import Wav2Vec2FeatureExtractor, Wav2Vec2Processor
# Load the preprocessor configuration from the local file
with open(local_preprocessor_config_path, 'r') as file:
preprocessor_config = json.load(file)
# Initialize the preprocessor using the loaded configuration
feature_extractor = Wav2Vec2FeatureExtractor(
do_normalize=preprocessor_config["do_normalize"],
feature_extractor_type=preprocessor_config["feature_extractor_type"],
feature_size=preprocessor_config["feature_size"],
padding_side=preprocessor_config["padding_side"],
padding_value=preprocessor_config["padding_value"],
processor_class_from_name=preprocessor_config["processor_class"],
return_attention_mask=preprocessor_config["return_attention_mask"],
sampling_rate=preprocessor_config["sampling_rate"]
)
# load the newly finetuned model from huggingface repo
from huggingface_hub import hf_hub_download
model_path = hf_hub_download(
repo_id="kvilla/wav2vec-english-speech-emotion-recognition-finetuned",
filename="model_finetuned.pth"
)
# load the newly finetuned model! from local
saved_model = torch.load(model_path, map_location=torch.device('cpu'))
# Create the model with the loaded configuration
model = Wav2Vec2ForSpeechClassification(config=config)
# Load the state dictionary
model.load_state_dict(saved_model)
print("Model initialized successfully.")
model.eval()
def recognize_emotion(audio):
# Load the audio file using librosa
sample_rate, audio_data = audio
# Ensure audio data is in floating-point format
if not np.issubdtype(audio_data.dtype, np.floating):
audio_data = audio_data.astype(np.float32)
# If you still want to process it with librosa, e.g., to change sample rate:
if sample_rate != 16000:
print(datetime.now().strftime('%Y-%m-%d%H:%M:%S'), ":Resampling audio...")
audio_data = librosa.resample(audio_data, orig_sr=sample_rate, target_sr=16000)
emotion, probabilities = predict(model, feature_extractor, audio_data, 48000, id2label) # limit to 3seconds
print(probabilities)
probs = probabilities.detach().numpy().flatten().tolist()
print(probs)
# Convert probabilities to percentages
percentages = [round(prob * 100, 2) for prob in probs]
print(percentages)
# Define the class labels (adjust to match your specific model's class labels)
labels = ["angry", "disgust", "fear", "happy", "neutral", "sad", "surprise"]
print(labels)
# Create a DataFrame
df = pd.DataFrame({"Emotion": labels, "Probability (%)": percentages})
df = df.sort_values(by="Probability (%)", ascending=False)
print(datetime.now().strftime('%Y-%m-%d%H:%M:%S'), df)
return emotion, get_emotion_image(emotion), df
def get_emotion_image(emotion):
# Here, you would have a dictionary or logic to map emotions to images
emotion_to_image = {
"angry": "angry.jpeg",
"disgust": "disgust.jpeg",
"fear": "fear.jpeg",
"happy": "happy.jpeg",
"neutral": "neutral.jpeg",
"sad": "sad.jpeg",
"surprise": "surprise.jpeg"
# Add other emotions and their corresponding images
}
# Default image if emotion is not found
image_path = emotion_to_image.get(emotion, "default.jpg")
# Load and return the image
return Image.open(image_path)
demo = gr.Blocks()
with demo:
df_logs = pd.DataFrame(columns=['Timestamp', 'Emotion'])
theme= gr.themes.Soft(),
audio_input = gr.Audio(type="numpy",
sources=["microphone"],
show_label=True,
streaming=True
)
text_output = gr.Textbox(label="Recognized Emotion")
output_df = gr.DataFrame(label="Emotion Probabilities")
image_output = gr.Image(label="Emotion Image", scale = 1, interactive = False)
df_logs = gr.DataFrame(label="Output Logs", headers = ['Timestamp', 'Emotion'])
def process_audio(audio, emotion, image, state, df_probs, df_logs):
current_time = time.time()
if state is None or (current_time - state >= 10):
state = current_time
emotion, image, df_probs = recognize_emotion(audio)
# Sample prediction data
timestamp = datetime.now().strftime('%Y-%m-%d %H:%M:%S')
# Create a dictionary for the new row
new_row = {'Timestamp': timestamp, 'Emotion': emotion}
# Append the new row to the DataFrame
df_logs = pd.concat([df_logs, pd.DataFrame([new_row])], ignore_index=True)
print(datetime.now().strftime('%Y-%m-%d%H:%M:%S'), "Predicted emotion: ", emotion)
return emotion, image, state, df_probs, df_logs
else:
print(datetime.now().strftime('%Y-%m-%d%H:%M:%S'), "Not yet time")
return emotion, image, state, df_probs, df_logs
# Automatically call the recognize_em otion function when audio is recorded
state = gr.State(None)
audio_input.stream(fn=process_audio, inputs=[audio_input, text_output, image_output, state, output_df, df_logs], outputs=[text_output, image_output, state, output_df, df_logs])
demo.launch(share=True)
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