Speech Signal Research Articles

Perception of vocoded speech in domestic dogs.

Humans have an impressive ability to comprehend signal-degraded speech; however, the extent to which comprehension of degraded speech relies on human-specific features of speech perception vs. more general cognitive processes is unknown. Since dogs live alongside humans and regularly hear speech, they can be used as a model to differentiate between these possibilities. One often-studied type of degraded speech is noise-vocoded speech (sometimes thought of as cochlear-implant-simulation speech). Noise-vocoded speech is made by dividing the speech signal into frequency bands (channels), identifying the amplitude envelope of each individual band, and then using these envelopes to modulate bands of noise centered over the same frequency regions - the result is a signal with preserved temporal cues, but vastly reduced frequency information. Here, we tested dogs' recognition of familiar words produced in 16-channel vocoded speech. In the first study, dogs heard their names and unfamiliar dogs' names (foils) in vocoded speech as well as natural speech. In the second study, dogs heard 16-channel vocoded speech only. Dogs listened longer to their vocoded name than vocoded foils in both experiments, showing that they can comprehend a 16-channel vocoded version of their name without prior exposure to vocoded speech, and without immediate exposure to the natural-speech version of their name. Dogs' name recognition in the second study was mediated by the number of phonemes in the dogs' name, suggesting that phonological context plays a role in degraded speech comprehension.

Prelingually Deaf Children With Cochlear Implants Show Better Perception of Voice Cues and Speech in Competing Speech Than Postlingually Deaf Adults With Cochlear Implants.

Postlingually deaf adults with cochlear implants (CIs) have difficulties with perceiving differences in speakers' voice characteristics and benefit little from voice differences for the perception of speech in competing speech. However, not much is known yet about the perception and use of voice characteristics in prelingually deaf implanted children with CIs. Unlike CI adults, most CI children became deaf during the acquisition of language. Extensive neuroplastic changes during childhood could make CI children better at using the available acoustic cues than CI adults, or the lack of exposure to a normal acoustic speech signal could make it more difficult for them to learn which acoustic cues they should attend to. This study aimed to examine to what degree CI children can perceive voice cues and benefit from voice differences for perceiving speech in competing speech, comparing their abilities to those of normal-hearing (NH) children and CI adults. CI children's voice cue discrimination (experiment 1), voice gender categorization (experiment 2), and benefit from target-masker voice differences for perceiving speech in competing speech (experiment 3) were examined in three experiments. The main focus was on the perception of mean fundamental frequency (F0) and vocal-tract length (VTL), the primary acoustic cues related to speakers' anatomy and perceived voice characteristics, such as voice gender. CI children's F0 and VTL discrimination thresholds indicated lower sensitivity to differences compared with their NH-age-equivalent peers, but their mean discrimination thresholds of 5.92 semitones (st) for F0 and 4.10 st for VTL indicated higher sensitivity than postlingually deaf CI adults with mean thresholds of 9.19 st for F0 and 7.19 st for VTL. Furthermore, CI children's perceptual weighting of F0 and VTL cues for voice gender categorization closely resembled that of their NH-age-equivalent peers, in contrast with CI adults. Finally, CI children had more difficulties in perceiving speech in competing speech than their NH-age-equivalent peers, but they performed better than CI adults. Unlike CI adults, CI children showed a benefit from target-masker voice differences in F0 and VTL, similar to NH children. Although CI children's F0 and VTL voice discrimination scores were overall lower than those of NH children, their weighting of F0 and VTL cues for voice gender categorization and their benefit from target-masker differences in F0 and VTL resembled that of NH children. Together, these results suggest that prelingually deaf implanted CI children can effectively utilize spectrotemporally degraded F0 and VTL cues for voice and speech perception, generally outperforming postlingually deaf CI adults in comparable tasks. These findings underscore the presence of F0 and VTL cues in the CI signal to a certain degree and suggest other factors contributing to the perception challenges faced by CI adults.

A speech enhancement model based on noise component decomposition: Inspired by human cognitive behavior

In multimedia intelligent systems, speech enhancement is commonly employed to improve the quality of speech signals, making them clearer and more natural. Current deep learning-based speech enhancement models typically treat noise as a unified entity and aim to separate it from the target speech. In this paper, inspired by the cognitive behavior of the human brain when observing noisy speech spectrograms, we decompose the spectral energy of noise into regular and random components. We propose an auxiliary-model-based speech enhancement framework that better suppresses noise components closely resembling speech features. Firstly, we introduce a voiceprint segmentation network (VSnet) that partitions noisy speech into voiceprint and non-voiceprint regions. Subsequently, we present a noise reconstruction network (NRnet) that utilizes noise information from non-voiceprint regions to reconstruct and suppress the regular noise components within the voiceprint region. Finally, we construct a combination of a model dedicated to suppressing random components (RANnet) and a speech enhancement model (SEnet), and train them synchronously. By sharing encoder parameters, SEnet is compelled to reduce the extraction of regular noise features from the original noisy speech, contributing to improving speech quality generated through the decoder. Experimental results on public Voickbank-DEMAND and DNS-challenge 2020 datasets demonstrate that our approach achieves state-of-the-art performance.

Effect of Face Masks on Automatic Speech Recognition Accuracy for Mandarin

Automatic speech recognition (ASR) has been widely used to realize daily human–machine interactions. Face masks have become everyday wear in our post-pandemic life, and speech through masks may have impaired the ASR. This study explored the effects of different kinds of face masks (e.g., surgical mask, KN95 mask, and cloth mask) on the Mandarin word accuracy of two ASR systems with or without noises. A mouth simulator was used to play speech audio with or without wearing a mask. Acoustic signals were recorded at distances of 0.2 m and 0.6 m. Recordings were mixed with two noises at a signal-to-noise ratio of +3 dB: restaurant noise and speech-shaped noise. Results showed that masks did not affect ASR accuracy without noise. Under noises, masks did not significantly influence ASR accuracy at 0.2 m but had significant effects at 0.6 m. The activated-carbon mask had the most significant impact on ASR accuracy at 0.6 m, reducing the accuracy by 18.5 percentage points compared to that without a mask, whereas the cloth mask had the least effect on ASR accuracy at 0.6 m, reducing the accuracy by 0.9 percentage points. The acoustic attenuation of masks on the high-frequency band at around 3.15 kHz of the speech signal attributed to the effects of masks on ASR accuracy. When training ASR models, it may be important to consider mask robustness.

Supervised Attention Multi-Scale Temporal Convolutional Network for monaural speech enhancement

Speech signals are often distorted by reverberation and noise, with a widely distributed signal-to-noise ratio (SNR). To address this, our study develops robust, deep neural network (DNN)-based speech enhancement methods. We reproduce several DNN-based monaural speech enhancement methods and outline a strategy for constructing datasets. This strategy, validated through experimental reproductions, has effectively enhanced the denoising efficiency and robustness of the models. Then, we propose a causal speech enhancement system named Supervised Attention Multi-Scale Temporal Convolutional Network (SA-MSTCN). SA-MSTCN extracts the complex compressed spectrum (CCS) for input encoding and employs complex ratio masking (CRM) for output decoding. The supervised attention module, a lightweight addition to SA-MSTCN, guides feature extraction. Experiment results show that the supervised attention module effectively improves noise reduction performance with a minor increase in computational cost. The multi-scale temporal convolutional network refines the perceptual field and better reconstructs the speech signal. Overall, SA-MSTCN not only achieves state-of-the-art speech quality and intelligibility compared to other methods but also maintains stable denoising performance across various environments.

Improving Speech Recognition with Convolutional Neural Networks

This project explores advanced techniques in speech recognition, focusing on emotion identification using Convolutional Neural Networks for improved accuracy and real-time processing efficiency. Emotion recognition from speech signals plays a crucial role in various applications, including human-computer interaction, customer service, mental health monitoring, and entertainment. This project proposes an innovative approach to emotion recognition using Convolutional Neural Networks (CNNs) applied to speech data. By leveraging advanced deep learning techniques, the proposed system aims to accurately identify and classify emotions conveyed through vocal expressions. The project begins with a comprehensive review of existing literature on emotion recognition and speech processing, identifying key challenges and opportunities in the field. Building upon prior research, the project introduces a novel CNN architecture optimized for emotion recognition tasks. This architecture is designed to extract relevant features from speech signals and capture subtle nuances indicative of different emotional states. One of the distinguishing features of the proposed approach is its multi-modal integration, which combines information from both audio and visual modalities to enhance emotion recognition accuracy. In addition to analysing speech signals, the system incorporates visual cues such as facial expressions and gestures, providing a more comprehensive understanding of the speaker's emotional state. Real-time processing efficiency is prioritized in the design of the system, ensuring prompt and responsive emotion recognition in interactive applications. Optimization techniques such as model quantization and lightweight architecture design are employed to minimize computational overhead while maintaining high accuracy. To address the variability and subjectivity of emotional expression, the system incorporates user-specific adaptation mechanisms. Through continuous learning and feedback integration, the system dynamically adapts to individual speakers' speech patterns and emotional characteristics, enhancing its ability to accurately recognize emotions in diverse contexts. The project also explores ensemble learning strategies to improve robustness and generalization performance. By combining predictions from multiple CNN models trained on diverse datasets, the system achieves greater resilience to variations in emotional expression and environmental factors. Ethical considerations, including privacy protection and responsible data handling, are integral aspects of the project's design and implementation. Measures are implemented to ensure the ethical collection, storage, and usage of speech data, safeguarding user privacy and maintaining trust in the system. Overall, the proposed system represents a significant advancement in emotion recognition technology, offering a sophisticated and versatile solution for accurately identifying emotions from speech signals. By leveraging deep learning techniques, multi-modal integration, real-time processing optimization, user-specific adaptation, and ensemble learning, the system demonstrates promising potential for various practical applications requiring robust and context-aware emotion recognition capabilities. Keywords: Speech Recognition, Emotion Identification, Convolutional Neural Networks (CNNs), Real-time Processing, Multi-modal Integration, User-specific Adaptation, Ensemble Learning, Deep Learning, Emotional Expression, Ethical Data Handling

Exploring the Impact of Mismatch Conditions, Noisy Backgrounds, and Speaker Health on Convolutional Autoencoder-Based Speaker Recognition System with Limited Dataset

This paper presents a novel approach to enhance the success rate and accuracy of speaker recognition and identification systems. The methodology involves employing data augmentation techniques to enrich a small dataset with audio recordings from five speakers, covering both male and female voices. Python programming language is utilized for data processing, and a convolutional autoencoder is chosen as the model. Spectrograms are used to convert speech signals into images, serving as input for training the autoencoder. The developed speaker recognition system is compared against traditional systems relying on the MFCC feature extraction technique. In addition to addressing the challenges of a small dataset, the paper explores the impact of a "mismatch condition" by using different time durations of the audio signal during both training and testing phases. Through experiments involving various activation and loss functions, the optimal pair for the small dataset is identified, resulting in a high success rate of 92.4% in matched conditions. Traditionally, Mel-Frequency Cepstral Coefficients (MFCC) have been widely used for this purpose. However, the COVID-19 pandemic has drawn attention to the virus's impact on the human body, particularly on areas relevant to speech, such as the chest, throat, vocal cords, and related regions. COVID-19 symptoms, such as coughing, breathing difficulties, and throat swelling, raise questions about the influence of the virus on MFCC, pitch, jitter, and shimmer features. Therefore, this research aims to investigate and understand the potential effects of COVID-19 on these crucial features, contributing valuable insights to the development of robust speaker recognition systems.

Stacked U-Net with Time–Frequency Attention and Deep Connection Net for Single Channel Speech Enhancement

Deep neural networks have significantly promoted the progress of speech enhancement technology. However, a great number of speech enhancement approaches are unable to fully utilize context information from various scales, hindering performance enhancement. To tackle this issue, we introduce a method called TFADCSU-Net (Stacked U-Net with Time-Frequency Attention (TFA) and Deep Connection Layer (DCL)) for enhancing noisy speech in the time–frequency domain. TFADCSU-Net adopts an encoder-decoder structure with skip links. Within TFADCSU-Net, a multiscale feature extraction layer (MSFEL) is proposed to effectively capture contextual data from various scales. This allows us to leverage both global and local speech features to enhance the reconstruction of speech signals. Moreover, we incorporate deep connection layer and TFA mechanisms into the network to further improve feature extraction and aggregate utterance level context. The deep connection layer effectively captures rich and precise features by establishing direct connections starting from the initial layer to all subsequent layers, rather than relying on connections from earlier layers to subsequent layers. This approach not only enhances the information flow within the network but also avoids a significant rise in computational complexity as the number of network layers increases. The TFA module consists of two attention branches operating concurrently: one directed towards the temporal dimension and the other towards the frequency dimension. These branches generate distinct forms of attention — one for identifying relevant time frames and another for selecting frequency wise channels. These attention mechanisms assist the models in discerning “where” and “what” to prioritize. Subsequently, the TA and FA branches are combined to produce a comprehensive attention map in two dimensions. This map assigns specific attention weights to individual spectral components in the time–frequency representation, enabling the networks to proficiently capture the speech characteristics in the T-F representation. The results confirm that the proposed method outperforms other models in terms of objective speech quality as well as intelligibility.

Predicting UPDRS in Parkinson’s disease using ensembles of self-organizing map and neuro-fuzzy

Parkinson's Disease (PD) is a complex, degenerative disease that affects nerve cells that are responsible for body movement. Artificial Intelligence (AI) algorithms are widely used to diagnose and track the progression of this disease, which causes symptoms of Parkinson's disease in its early stages, by predicting the results of the Unified Parkinson's Disease Rating Scale (UPDRS). In this study, we aim to develop a method based on the integration of two methods, one complementary to the other, Ensembles of Self-Organizing Map and Neuro-Fuzzy, and an unsupervised learning algorithm. The proposed method relied on the higher effect of the variables resulting from the analysis of the initial readings to obtain a correct and accurate preliminary prediction. We evaluate the developed approach on a PD dataset including speech cues. The process was evaluated with root mean square error (RMSE) and modified R square (modified R2). Our findings reveal that the proposed method is effective in predicting UPDRS outcomes by a combination of speech signals (measures of hoarseness). As the preliminary results during the evaluation showed numbers that proved the worth of the proposed method, such as UPDRS = 0.955 and RMSE approximately 0.2769 during the prediction process.

Method for asynchronous analysis of a glottal source based on a two-level autoregressive model of the speech signal

The task of analyzing a glottal source over a short observation interval is considered. The acute problem of insufficient performance of known methods for analyzing a glottal source is pointed out, regardless of the mode of data preparation: synchronous with the main tone of speech sounds or asynchronous. A method for analyzing the glottal source based on a two-level autoregressive model of the speech signal is proposed. Its software implementation based on the high-speed Burg-Levinson computational procedure is described. It does not require synchronization of the sequence of observations used with the main tone of the speech signal and is characterized by a relatively small amount of computational costs. Using the described software implementation, a full-scale experiment was set up and conducted, where the vowel sounds of the control speaker’s speech were used as the object of study. Based on the results of the experiment, the increased performance of the proposed method was confirmed and its requirements for the duration of the speech signal during voice analysis in real time were formulated. It is shown that the optimal duration is in the range from 32 to 128 ms. The results obtained can be used in the development and research of digital speech communication systems, voice control, biometrics, biomedicine and other speech systems where the voice characteristics of the speaker’s speech are of paramount importance.

Emotion Modeling in Speech Signals: Discrete Wavelet Transform and Machine Learning Tools for Emotion Recognition System

Speech emotion recognition (SER) is a challenging task due to the complex and subtle nature of emotions. This study proposes a novel approach for emotion modeling using speech signals by combining discrete wavelet transform (DWT) with linear prediction coding (LPC). The performance of various classifiers, including support vector machine (SVM), K-Nearest Neighbors (KNN), Efficient Logistic Regression, Naive Bayes, Ensemble, and Neural Network, was evaluated for emotion classification using the EMO-DB dataset. Evaluation metrics such as area under the curve (AUC), average prediction accuracy, and cross-validation techniques were employed. The results indicate that KNN and SVM classifiers exhibited high accuracy in distinguishing sadness from other emotions. Ensemble methods and Neural Networks also demonstrated strong performance in sadness classification. While Efficient Logistic Regression and Naive Bayes classifiers showed competitive performance, they were slightly less accurate compared to other classifiers. Furthermore, the proposed feature extraction method yielded the highest average accuracy, and its combination with formants or wavelet entropy further improved classification accuracy. On the other hand, Efficient Logistic Regression exhibited the lowest accuracies among the classifiers. The uniqueness of this study was that it investigated a combined feature extraction method and integrated them to compare with various forms of combinations. However, the purposes of the investigation include improved performance of the classifiers, high effectiveness of the system, and the potential for emotion classification tasks. These findings can guide the selection of appropriate classifiers and feature extraction methods in future research and real-world applications. Further investigations can focus on refining classifiers and exploring additional feature extraction techniques to enhance emotion classification accuracy.

Using transformers for multimodal emotion recognition: Taxonomies and state of the art review

Emotion recognition is an aspect of human-computer interaction, affective computing, and social robotics. Conventional unimodal approaches for emotion recognition, depending on single data sources such as facial expressions or speech signals often fall short in capturing the complexity and context-dependent nature of emotions. Multimodal Emotion Recognition (MER), which integrates information from multiple modalities, has emerged as a promising solution to overcome these limitations. In recent years, Transformers-based approaches have gathered significant attention in the fields of natural language processing and computer vision, highlighting their ability to capture long-range dependencies and semantic representations. These models have rapidly achieved the MER state-of-the-art. However, current survey papers that cover MER lack a specific focus on Transformer-based techniques. To bridge this research gap, this review paper provides a comprehensive investigation of Transformers-based approaches for MER. It explores various Transformer architectures and proposes several scenarios for using Transformers at different stages of MER process. In addition, it examines datasets suitable for MER, discusses fusion mechanisms, and introduces novel taxonomies in both MER and Transformer technologies. The review also addresses challenges and future research directions. Through this review, we aim to provide researchers with an inclusive understanding of the current state-of-the-art in Transformers-based approaches for MER, paving the way for further advancements in this rapidly developing field.

Comparative Performance Analysis of Metaheuristic Feature Selection Methods for Speech Emotion Recognition

Abstract Emotion recognition systems from speech signals are realized with the help of acoustic or spectral features. Acoustic analysis is the extraction of digital features from speech files using digital signal processing methods. Another method is the analysis of time-frequency images of speech using image processing. The size of the features obtained by acoustic analysis is in the thousands. Therefore, classification complexity increases and causes variation in classification accuracy. In feature selection, features unrelated to emotions are extracted from the feature space and are expected to contribute to the classifier performance. Traditional feature selection methods are mostly based on statistical analysis. Another feature selection method is the use of metaheuristic algorithms to detect and remove irrelevant features from the feature set. In this study, we compare the performance of metaheuristic feature selection algorithms for speech emotion recognition. For this purpose, a comparative analysis was performed on four different datasets, eight metaheuristics and three different classifiers. The results of the analysis show that the classification accuracy increases when the feature size is reduced. For all datasets, the highest accuracy was achieved with the support vector machine. The highest accuracy for the EMO-DB, EMOVA, eNTERFACE’05 and SAVEE datasets is 88.1%, 73.8%, 73.3% and 75.7%, respectively.

Speech Emotion Recognition using Hybrid Architectures

The detection of human emotions from speech signals remains a challenging frontier in audio processing and human-computer interaction domains. This study introduces a novel approach to Speech Emotion Recognition (SER) using a Dendritic Layer combined with a Capsule Network (DendCaps). A Convolutional Neural Network (NN) and a Long Short-Time Neural Network (CLSTM) hybrid model are used to create a baseline which is then compared to the DendCap model. Integrating dendritic layers and capsule networks for speech emotion detection can harness the unique advantages of both architectures, potentially leading to more sophisticated and accurate models. Dendritic layers, inspired by the nonlinear processing properties of dendritic trees in biological neurons, can handle the intricate patterns and variabilities inherent in speech signals, while capsule networks, with their dynamic routing mechanisms, are adept at preserving hierarchical spatial relationships within the data, enabling the model to capture more refined emotional subtleties in human speech. The main motivation for using DendCaps is to bridge the gap between the capabilities of biological neural systems and artificial neural networks. This combination aims to capitalize on the hierarchical nature of speech data, where intricate patterns and dependencies can be better captured. Finally, two ensemble methods namely stacking and boosting are used for evaluating the CLSTM and DendCaps networks and the experimental results show that stacking of the CLSTM and DendCaps networks gives the superior result with a 75% accuracy.

Automatic speaker and age identification of children from raw speech using sincNet over ERB scale

This paper presents the newly developed non-native children’s English speech (NNCES) corpus to reveal the findings of automatic speaker and age recognition from raw speech. Convolutional neural networks (CNN), which have the ability to learn low-level speech representations, can be fed directly with raw speech signals instead of using traditional hand-crafted features. Moreover, the filters that were learned using standard CNNs appeared to be noisy because they consider all elements of each filter. In contrast, sincNet can be able to generate more meaningful filters simply by replacing the first convolutional layer by a sinc-layer in standard CNNs. The low and high cutoff frequencies of the rectangular band-pass filter are the only parameters that can be learned in sincNet, which has the potential to extract significant speech cues from the speaker, such as pitch and formants. In this work, the sincNet model is significantly changed by switching from baseline Mel scale initializations to equivalent rectangular bandwidth (ERB) initializations, which has the added benefit of allocating additional filters in the lower region of the spectrum. Additionally, it needs to be highlighted that the novel sincNet model is well suited to identify the age of the children. The investigations on both read and spontaneous speech tasks in speaker identification, gender independent & dependent age-group identification of children outperform the baseline models with varying relative improvements in terms of accuracy.

An attack-agnostic defense method against adversarial attacks on speaker verification by fusing downsampling and upsampling of speech signals

With the advance of deep learning, adversarial attack and defense has becoming a hot research topic. However, existing defense methods rely on the prior knowledge of the adversarial attacks, and are also vulnerable to adaptive attacks. In this paper, towards a secure automatic speaker verification system, a novel attack-agnostic defense method is proposed, named by SampleShield, which consists of a pairwise random downsampling (PRD) module and an upsampling module achieved by speech super-resolution (SSR). The PRD module can destroy the adversarial perturbations by downsampling, which also introduces randomness and non-differentiability, leading to its resistance to adaptive attacks. PRD does not use any adversarial example as training data, which makes it an attack-agnostic defense method with good generalization ability. Furthermore, the upsampling module achieved by a neural network can recover the downsampled speech to its original quality. In summary, given the experimental results and analysis on the benchmark public datasets, by fusing the downsampling and upsampling modules, SampleShield achieves excellent performance. On adversarial defense, SampleShield obtains an error rate about 5.2% which is quite close to the idea lower bound value 3.2%. On speech quality, SampleShield gets a 5 dB signal-to-noise ratio improvement with respect to the best existing method.

Smart Hand Gloves for Deaf and Dumb People

Abstract: Approximately 2.78% of the population in our nation is nonverbal. They communicate with people by using their hands and facial gestures. As is well known, those who are silent are mute and those who are not are unaware of sign language, which is a way for silent individuals to communicate. Thus, we have put forth a brand-new method known as “The SPEAKING SYSTEM TO THE MUTE PEOPLE”. Thus, this system will be helpful in solving the issue. The accelerometer sensors form the foundation of the system. Mute people believe that every deed has a purpose. The accelerometer sensors accelerate with each action and provide a signal to the Arduino. The speech signal is created by the Arduino by synchronizing its motion with that of the database. The Guide Me app is being used as the system's output. Since gestures are consistent with people's communication habits, a lot of study has been done on gesture detection using a vision-based method.

Comparative Analysis Using Machine Learning Algorithms to Detect Parkinson’s Disease using Voice Dataset

Abstract: Parkinson's disease is a degenerative disorder that affects the control of movement, and its symptoms include tremors, stiffness, and difficulties in balance and coordination. This disease results from a decrease in dopamine-producing cells in the brain, and although there is currently no cure for Parkinson's, medication and therapeutic interventions can help alleviate the symptoms. Parkinson's disease usually begins to show its effects in middle to late adulthood. A new measure of dysphonia, Pitch Period Entropy (PPE) assesses the variability or randomness of pitch periods in speech signals, providing insights into the diversity of vocal characteristics. Higher entropy values suggest greater pitch variability in a speaker's voice. PPE is included in the dataset. The methodologies used in this project are Random Forest, Support vector classifier, Extreme gradient boosting (XGB). This comparative study includes performance metrics: Accuracy, precision, Recall, f1 score, and support. The aim is to predict Parkinson's disease and compare performance metrics using various methods.

An Innovative Method for Speech Signal Emotion Recognition Based on Spectral Features Using GMM and HMM Techniques

An Innovative Method for Speech Signal Emotion Recognition Based on Spectral Features Using GMM and HMM Techniques

Exploring the Potential of Deep Learning in the Classification and Early Detection of Parkinson's Disease

INTRODUCTION: Parkinson's Disease (PD) is a progressive neurological disorder affecting a significant portion of the global population, leading to profound impacts on daily life and imposing substantial burdens on healthcare systems. Early identification and precise classification are crucial for effectively managing this disease. This research investigates the potential of deep learning techniques in facilitating early recognition and accurate classification of PD.
 OBJECTIVES: The primary objective of this study is to leverage advanced deep learning techniques for the early detection and precise classification of Parkinson's Disease. By utilizing a rich dataset comprising speech signal features extracted from 3000 PD patients, including Time Frequency Features, Mel Frequency Cepstral Coefficients (MFCCs), Wavelet Transform based Features, Vocal Fold Features, and TWQT features, this research aims to evaluate the performance of various deep learning models in PD classification.
 METHODS: The dataset containing diverse speech signal features from PD patients' recordings serves as the foundation for training and evaluating five different deep learning models: ResNet50, VGG16, Inception v2, AlexNet, and VGG19. Each model undergoes training and assessment to determine its capability in accurately classifying PD patients. Performance metrics such as accuracy are employed to evaluate the models' effectiveness.
 RESULTS: The results demonstrate promising potential, with overall accuracies ranging from 89% to 95% across the different deep learning models. Notably, AlexNet emerges as the top-performing model, achieving an accuracy of 95% and demonstrating balanced performance in accurately identifying both true and false PD cases.
 CONCLUSION: This research highlights the significant potential of deep learning in facilitating the early detection and classification of Parkinson's Disease. Leveraging speech signal features offers a non-invasive and cost-effective approach to PD assessment. The findings contribute to the growing body of evidence supporting the integration of artificial intelligence in healthcare, particularly in the realm of neurodegenerative disorders. Further exploration into the application of deep learning in this domain holds promise for advancing PD diagnosis and management.