Journal of the Audio Engineering Society

2002 July/August - Volume 50 Number 7/8

Papers


An algorithm for the correction of disturbances or gaps of up to several thousand samples in an audio signal is presented. The reconstruction is based on a novel method for time-domain discrete signal extrapolation. The missing or disturbed portion of the audio signal is replaced by a weighted average of signals extrapolated from the areas preceding and following the disturbed portion. Impulsive-type errors usually distort the underlying signal irreversibly, and the damaged signal portion does not contain any information of the original signal. In the proposed method the damaged signal samples are not used in computing the replacing samples. The reconstruction method is applied in practice to correct scratches from signals recorded from badly damaged vinyl recordings. The proposed signal reconstruction method can be implemented in real-time applications.

In part I of two papers the requirements for low-frequency sound reproduction were investigated by the variation of lower cutoff frequency and slope and by the introduction of different levels of amplitude ripple and group delay ripple in the passband of a high-performance sound reproduction system. Listening tests were performed at three different sound pressure levels using both loudspeakers in an anechoic chamber and headphones in an audiometric booth. Two reproduction setups were used to confirm that equal results of the listening tests could be obtained in the two cases when proper equalization was implemented. It is described how DSP was used to generate stimuli and perform equalization of the two reproduction setups. The shape and magnitude of amplitude and group delay ripple were derived from room simulations of an IEC 268-13 sized room with varying reverberation time. Proper equalization included the introduction of head-related transfer functions in the signal path to the headphones. This ensured that the sound pressures at the ear drums were very similar in two cases: a person sitting in front of the loudspeakers in the anechoic chamber and a person wearing headphones in the experimental booth. Level calibration was performed on both setups using pink noise. The nonlinearities measured in the physical loudspeakers were introduced into the signal path to the headphones using a nonlinearity simulator program.

The audibility of changes in passband amplitude ripple and lower system cutoff frequency and slope has been investigated for a loudspeaker system for two situations: a real loud-speaker in an anechoic chamber and a simulated system reproduced via headphones. The signals were standard program material, selected to ensure a sufficient energy content at the relevant frequencies. The experiments were conducted with six subjects with normal hearing using a paired-comparison procedure. The subjects assessed the attributes "lower bass" and "upper bass" in relation to a fixed reference condition. The first experiment investigated the influence of high-pass filter order (second, fourth, and sixth) and lower cutoff frequency (20, 35, and 50 Hz) at three reproduction levels and for four program items. The second experiment examined the influence of amplitude ripple corresponding to four reverberation times at three reproduction levels and for four program items. The results of the first experiment showed that the lower cutoff frequency has a significant influence on the perceived level of lower bass reproduction if the reproduction level is above the hearing threshold in the relevant frequency bands. The influence of high-pass filter order was not significant for the conditions investigated. The results of the second experiment showed that the amplitude ripple has a significant influence on the perceived level of lower and upper bass reproduction. The results also showed that there were no significant differences between the data produced by the two reproduction methods.

On the Design and Efficiency of Class A, B, AB, G, and H Audio Power Amplifier Output Stages

Authors: Bortoni, Rosalfonso; Filho, Sidnei Noceti; Seara, Rui

A procedure for analyzing, designing, and assessing audio power amplifier output stages operating in classes A, B, AB, G, and H with reactive loads is presented. The study considers steady-state sinusoidal analysis for BJT, IGBT, and MOSFET technologies. Electrical/mechanical/acoustical models of loudspeakers and enclosures are used whose parameters are obtained through the Thiele/Small model. An equivalent electrical thermal model for a transistor heatsink ambience system associated with instantaneous and average powers is used for designing the power stage. A MATLAB software has been developed, which provides considerable support to the designer for all phases required in the design of audio power amplifier output stages.

Engineering reports


Requirements for Low-Frequency Sound Reproduction, Part II: Generation of Stimuli and Listening System Equalization

Reconstruction Method for Missing or Damaged Long Portions in Audio Signal

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