Jaywoo Kim

Adaptive Microphone Array System with Two-Stage Adaptation Mode Controller*This work was supported by the Research Fund of Samsung Advanced Institute of Technology under Project PR00002417RR.

In this paper, an adaptive microphone array system with a two-stage adaptation mode controller (AMC) is proposed for high-quality speech acquisition in real environments. The proposed system includes an adaptive array algorithm, a time-delay estimator and a newly proposed AMC. To ensure proper adaptation of the adaptive array algorithm, the proposed AMC uses not only temporal information, but also spatial information. The proposed AMC is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive array algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed algorithm is implemented as a real-time man-machine interface module of a home-agent robot. Simulation results show 13 dB SINR improvement with the speaker sitting 2 m distance from the home-agent robot. The speech recognition rate is also enhanced by 32% when compared to the single channel acquisition system.

Speech enhancement and recognition using circular microphone array for service robots

Speech recognition using circular microphone array is addressed in this paper. Eight microphones are located around the service robot to form a 2D microphone array. To enhance the speech quality, a novel adaptive beamformer composed of a delay-and-sum beamformer, adaptive blocking filters (ABFs) and adaptive cancelling filters (ACFs) is proposed. While the adaptive generalized sidelobe canceller (AGSC) connects the ABF and the ACF in feedforward, the proposed adaptive beamformer has them in feedback. The advantages of the proposed structure are the robustness to the steering vector errors and cross-talks and the reduced number of filter taps that gives the same speech quality compared to the AGSC with a larger number of filter taps. The experimental results show that the proposed structure is superior to the AGSC in objective and subjective evaluations. Speech recognition result shows that the proposed robust adaptive beamformer guarantees the recognition performance even in a low SNR and highly reverberant environment.

Heart Rate Monitor for Portable MP3 Player

This paper presents a photoplethysmography sensor based on a heart rate monitor for a portable MP3 player. Two major design issues are addressed: one is to acquire the sensor signal with a proper amplitude despite a wide range of variation and the other is to handle the noise contaminated signal which is caused by a motion artifact. A benchmarking test with a professional medical photoplethysmography sensor shows that our device performs very well in calculating heart rate even though our photoplethysmography sensor module was designed to be cost effective

Method of Measuring Calorie Consumption for Portable Apparatus

A measuring device for use in measuring calorie consumption includes a pulse input unit which detects a first heart rate of the user at a first point in an exercise period and a second heart rate of the user at a second point of the exercise period other than the first point, the pulse input unit not detecting a third heart rate between the first and second heart rates; and a controller which receives the detected first and second heart rates and calculates calories consumed using the detected first heart rate, the detected second heart rate, an at rest heart rate of the user, and one or more of an age, gender, weight, height and an at rest heart rate of the user. The measuring device is usable in a portable device, such as a portable digital audio and/or video reproducing apparatus

High-quality speech acquisition and recognition system for home-agent robot

A PC-based real-time speech processing system designed for the home-agent robot is presented. The key feature of the proposed system is the speech acquisition module using a microphone array which enables acquiring high-quality speech of a distant talker.