Leveraging Artificial Neural Networks for Real-Time Speech Recognition in Voice-Activated Systems

It has further shaped the domain of real-time voice-controlled speech recognition systems. Things like language bias, computational expense and background noise have made strides more difficult in the past. This paper provides a novel view on these tasks, allowing for broader accessibility and real-world applicability of state-of-the-art models. We advocate a multi-dimensional methodology, including ad hoc model contextualization, tailored neural designs, and personalized learning strategies, to achieve voice and chip-height optimization. Existing speech recognition systems have a major limitation of being limited to few languages, dialects and accents. This study introduces a multilingual and multicultural model to address this issue and also provides access to the benefit of no-bias technologies to all parts of society. Out of these regional filters, it can work with more and work on them more precisely. In addition to this, the efficient structure also improves computational efficiency, meaning that the model is able to gain speed in real-time processing on low-power devices, fitting to rising demand for speech recognition in mobile and edge computing settings. Performance in contextual understanding remains a problem, errors in pronunciation or deviations from the dialect tend to lead to mistakes. As a result, the current study utilise semantic analysis and natural language processing (NLP) methods to assist understanding across different languages globally. These services are used in applications like medical/legal transcription or customer support, where correct transcription is critical. Additionally, the architecture enhances real-time processing by reducing latency and increasing responsiveness, which is critical in emergency response systems and autonomous vehicles where timely decision-making is crucial. By enhancing the efficiency and accuracy of ANN-based speech recognition, this research drives advancements in increasingly more accessible, effective, and reliable voice-activated technologies.

Authors

Institutions

Publication Details

Journal
ITM Web of Conferences
Published
2025-01-01
DOI
https://doi.org/10.1051/itmconf/20257601003
Primary Topic
Speech Recognition and Synthesis
Type
article
Field-Weighted Citation Impact
0.00
Controls
|||
ALL TIME
JAN
FEB
MAR
APR
MAY
JUN
JUL
AUG
SEP
OCT
article

Leveraging Artificial Neural Networks for Real-Time Speech Recognition in Voice-Activated Systems

Anisetty Suresh Kumar, G. Merlin Sheeba, Arun Raj S R, Suresh Kumar et al.
ITM Web of Conferences
Speech Recognition and Synthesis
article

Leveraging Artificial Neural Networks for Real-Time Speech Recognition in Voice-Activated Systems

Anisetty Suresh Kumar, G. Merlin Sheeba, Arun Raj S R, Suresh Kumar, Shanmugam Sureshkumar, Raveendra Nadh B
article en

Abstract

It has further shaped the domain of real-time voice-controlled speech recognition systems. Things like language bias, computational expense and background noise have made strides more difficult in the past. This paper provides a novel view on these tasks, allowing for broader accessibility and real-world applicability of state-of-the-art models. We advocate a multi-dimensional methodology, including ad hoc model contextualization, tailored neural designs, and personalized learning strategies, to achieve voice and chip-height optimization. Existing speech recognition systems have a major limitation of being limited to few languages, dialects and accents. This study introduces a multilingual and multicultural model to address this issue and also provides access to the benefit of no-bias technologies to all parts of society. Out of these regional filters, it can work with more and work on them more precisely. In addition to this, the efficient structure also improves computational efficiency, meaning that the model is able to gain speed in real-time processing on low-power devices, fitting to rising demand for speech recognition in mobile and edge computing settings. Performance in contextual understanding remains a problem, errors in pronunciation or deviations from the dialect tend to lead to mistakes. As a result, the current study utilise semantic analysis and natural language processing (NLP) methods to assist understanding across different languages globally. These services are used in applications like medical/legal transcription or customer support, where correct transcription is critical. Additionally, the architecture enhances real-time processing by reducing latency and increasing responsiveness, which is critical in emergency response systems and autonomous vehicles where timely decision-making is crucial. By enhancing the efficiency and accuracy of ANN-based speech recognition, this research drives advancements in increasingly more accessible, effective, and reliable voice-activated technologies.

ITM Web of ConferencesVol. 76
National Institute of Technology Tiruchirappalli (IN), University B.D.T College of Engineering (IN), Birla Institute of Technology and Science - Hyderabad Campus (IN), Institute of Engineering (NP), Saveetha University (IN), Bharath University (IN)
Openalex Percentile: Top 98%
Speech Recognition and Synthesis
AI Navigator

Ask Laika to Summarize, Analyze, and Connect papers live on the map.

Summarize Papers & Methodologies

Extract key findings, datasets, and comparative methods across publications.

Benchmark Rankings & Visual Analytics

Rank top research institutions, authors, funders, topics, and journals by Field-Weighted Citation Impact (FWCI) and paper volume with instant charts.

Connect Distant Disciplines

Bridge topological clusters on the map to find hidden collaborative intersections.