Journal of the Audio Engineering Society

2003 December - Volume 51 Number 12


Correction to: `Effects on Down-Mix Algorithms on Quality of Surround Sound`

Authors: Zielinski, Slawomir K.; Rumsey, Francis; Bech, Søren

CORRECTION TO 'EFFECTS ON DOWN-MIX ALGORITHMS ON QUALITY OF SURROUND SOUND' In the above paper,1 three errors occurred in the Appendix (p. 796). They include the term rms power, the values given in decibels, and the term output power 400 W (8 ohm). In Section A.1 the text should have read as follows. The values presented in Table 7 are digital signal levels in decibels referred to digital full scale. In other words, Ld 20 log(as /afs) where Ld is the signal level in decibels, as is the digital signal amplitude, and afs is the digital signal full-scale amplitude. In Section A.2 the term output power 400 W (8 ohm) should not be interpreted as the acoustical power produced by the subwoofer but as the electrical output power of the subwoofer's amplifier delivered to the drive unit. 1S. K. Zielinski, F. Rumsey, and S. Bech, J. Audio Eng. Soc., vol. 51, pp. 780-798 (2003 Sept.).

Author's Reply to Letter to the editor

Letter to the editor

Audio information stored in the undulations of grooves in a medium such as a phonograph record may be reconstructed, with no or minimal contact, by measuring the groove shape using precision metrology methods and digital image processing. The effects of damage, wear, and contamination may be compensated, in many cases, through image processing and analysis methods. The speed and data-handling capacity of available computing hardware make this approach practical. Various aspects of this approach are discussed. A feasibility test is reported which used a general-purpose optical metrology system to study a 50-year-old 78- rpm phonograph record. Comparisons are presented with stylus playback of the record and with a digitally remastered version of the original magnetic recording. A more extensive implementation of this approach, with dedicated hardware and software, is considered.

The relationship between the duration of a sound presentation and the accuracy of human localization is investigated. The three-dimensional sound is presented via headphones. The head-tracking system was integrated together with the sound presentation. Generalized headrelated transfer functions (HRTFs) are used in the experiment. Six different types of sounds with durations of 0.5, 2, 4, and 6 seconds were presented in random order on any azimuth in the horizontal plane. Thirty subjects participated in the study. A special location indication system called DINC (directional indication compass) was developed. With DINC the judged location of every test can be recorded accurately. The results showed that the localization accuracy is significantly related to the duration of the sound presentation. As long as the sound has a broad frequency bandwidth, the sound type has little effect on the localization accuracy. A presentation of at least 4-second duration is recommended. There is no significant difference between male and female subjects in the accuracy of detection.

Smart Digital Loudspeaker Arrays

Authors: Hawksford, Malcolm J.

A theory of smart loudspeaker arrays is described where a modified Fourier technique yields complex filter coefficients to determine the broad-band radiation characteristics of a uniform array of micro drive units. Beamwidth and direction are individually programmable over a 180° arc, where multiple agile and steerable beams carrying dissimilar signals can be accommodated. A novel method of stochastic filter design is also presented, which endows the directional array with diffuse radiation properties.

The level of broadcast sound is usually limited to prevent overmodulation of the transmitted signal. To increase the loudness of broadcast sounds, especially commercials, fastacting amplitude compression is often applied. This allows the root-mean-square (rms) level of the sounds to be increased without exceeding the maximum permissible peak level. In addition, even for a fixed rms level, compression may have an effect on loudness. To assess whether this was the case, we obtained loudness matches between uncompressed speech (short phrases) and speech that was subjected to varying degrees of four-band compression. All rms levels were calculated off line. We found that the compressed speech had a lower rms level than the uncompressed speech (by up to 3 dB) at the point of equal loudness, which implies that, at equal rms level, compressed speech sounds louder than uncompressed speech. The effect increased as the rms level was increased from 50 to 65 to 80 dB SPL. For the largest amount of compression used here, the compression would allow about a 58% increase in loudness for a fixed peak level (equivalent to a change in level of about 6 dB). With a slight modification, the model of loudness described by Glasberg and Moore [1] was able to account accurately for the results.

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