Subband-based acoustic shock limiting algorithm on a low-resource DSP system

Acoustic Shock describes a condition where sudden loud acoustic signals in communication equipment causes hearing damage and discomfort to the users. To combat this problem, a subband-based acoustic shock limiting (ASL) algorithm is proposed and implemented on an ultra low-power DSP system with an input-output latency of 6.5 msec. This algorithm processes the input signal in both the time and frequency domains. This approach allows the algorithm to detect sudden increases in sound level (time-domain), as well as frequencyselectively suppressing shock disturbances in frequency domain. The unaffected portion of the sound spectrum is thus preserved as much as possible. A simple ASL algorithm calibration procedure is proposed to satisfy different sound pressure level (SPL) limit requirements for various communication equipment. Acoustic test results show that the ASL algorithm limits acoustic shock signals to below specified SPL limits while preserving speech quality.

[1]  Robert L. Brennan,et al.  A multichannel compression strategy for a digital hearing aid , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[2]  S. Haykin,et al.  Adaptive Filter Theory , 1986 .

[3]  R. Brennan,et al.  A flexible filterbank structure for extensive signal manipulations in digital hearing aids , 1998, ISCAS '98. Proceedings of the 1998 IEEE International Symposium on Circuits and Systems (Cat. No.98CH36187).

[4]  Schuyler Quackenbush,et al.  Objective measures of speech quality , 1995 .

[5]  Luzheng Lu A digital realization of audio dynamic range control , 1998, ICSP '98. 1998 Fourth International Conference on Signal Processing (Cat. No.98TH8344).

[6]  S. Biyiksiz,et al.  Multirate digital signal processing , 1985, Proceedings of the IEEE.