Developing an audio network system for teleconferencing with evaluation of audio quality
Hansol Lim, Hyung Suk Jang, Muhammad Imran, Jin Yong Jeon · The Journal of the Acoustical Society of America · 2015
A teleconference system was developed to transmit speech signals for audio communication with remote users. The hardware system was composed of 3-D microphone arrays to capture directional sound and a wireless headset to provide freedom of movement for users. An application programming interface (API) was used for interaction between the computer and the audio devices, including the microphone, headphone, and ADDA converter, transferring input/output audio signals in real time through the network. For the evaluation of the system, a latency test was performed with a simultaneous recording system in capturing and reproduction positions. The latency from the network procedure was calculated with different buffer sizes, and the lower-delay and smaller-deviation buffer was selected. In addition, the transmitted audio signals were compared with the input signals in terms of signal patterns and frequency responses. The resulting audio qualities in the time and frequency domain were suitable for a teleconferencing system.