Umudike Journal of Engineering and Technology

Text-to-speech (TTS) technology has become an integral part of modern digital systems, enabling smooth interaction between humans and computers in areas like accessibility, education, and content creation. Despite major advancements, many TTS systems still struggle with high latency, limited scalability, and poor integration in web environments. These issues make it hard to provide real-time, high-quality speech synthesis for active user applications. This study aims to design and implement an efficient, low-latency TTS system by integrating the Groq PlayAI-TTS model into a strong client-server setup. The main goal is to improve real-time speech generation while ensuring scalability, usability, and smooth communication between system parts. The methodology follows a clear design approach. A responsive frontend captures user input and sends it to a backend API (Application Programming Interface) for processing. The backend uses the PlayAI-TTS model to generate high-quality audio output, which then goes back to the frontend for immediate playback. The system is optimized through effective request handling, solid error management, and adaptive performance in different network conditions. Results from the implementation show significant gains in response time, audio quality, and system reliability compared to traditional TTS solutions. The system achieves near real-time speech synthesis with very little latency. It also supports multiple languages and customizable user profiles. Furthermore, its scalable design allows it to be used across various web platforms. Conclusively, the developed system offers a practical and high-performance solution for modern TTS applications. By connecting improved speech synthesis models with a user-friendly design, the study illustrates the potential of the Groq PlayAI-TTS system in providing efficient, accessible, and scalable speech technologies.