A personal voice analyzer and trainer

This paper presents a personal voice analyzer and trainer that allow the user to perform four daily exercises to improve the voice capacity. The system grades how well the user is performing the exercises by analyzing the duration, the intensity and the pitch of the user’s voice

An improved adaptive gain equalizer for noise reduction with low speech distortion

In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as possible. Previous work, based on time varying amplification controlled by signal-to-noise ratio estimation in different frequency subbands, has shown promising results in this regard but can suffer from problems in situations with intense continuous speech. Further, the amount of noise reduction cannot exceed a certain level in order to avoid artifacts. This paper establishes the problems and proposes several improvements. The improved algorithm is evaluated with several different noise characteristics, and the results show that the algorithm provides even less speech distortion, better performance in a multi-speaker environment and improved noise suppression when speech is absent compared with previous work.Open Access article Article 7</p

A Personal Voice Analyzer and Trainer

This paper presents a personal voice analyzer and trainer that allow the user to perform four daily exercises to improve the voice capacity. The system grades how well the user is performing the exercises by analyzing the duration, the intensity and the pitch of the user’s voice

An improved adaptive gain equalizer for noise reduction with low speech distortion

In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as possible. Previous work, based on time varying amplification controlled by signal-to-noise ratio estimation in different frequency subbands, has shown promising results in this regard but can suffer from problems in situations with intense continuous speech. Further, the amount of noise reduction cannot exceed a certain level in order to avoid artifacts. This paper establishes the problems and proposes several improvements. The improved algorithm is evaluated with several different noise characteristics, and the results show that the algorithm provides even less speech distortion, better performance in a multi-speaker environment and improved noise suppression when speech is absent compared with previous work.Open Access article Article 7</p

Low-complexity network echo cancellation approach for systems equipped with external memory

Long delays and sparseness characterize impulse responses in telecommunication networks and a vast number of solutions for network echo cancellation have been proposed over the years. In this paper, an approach for detecting dispersive regions of a sparse impulse response and a proportionate normalized least mean square (PNLMS)-based selective updating approach are combined with an adaptive double-talk detector to form a complete solution for echo cancellation. The proposed solution has low computational complexity and is targeted for systems equipped with external memory