In This Section
Journal of the AES
2012 December - Volume 60 Number 12
Editor’s Note and Reviewer List
When the linear and time-invariant assumptions are valid, measuring the impulse response of a system is an ideal way to capture its properties such that output with any input can be computed with convolution. Nonlinear systems with memory, for example musical instruments and vacuum tube, require an approach such as Volterra series. Measurement technique can capture the information of the kernels of the diagonal Volterra model for the system being measured. To demonstrate the efficiency of this theoretical formulation, a nonlinear device (Tube Screamer) was tested. The richness of its sound resides mainly in the harmonic Distortions.
To what extent does the mixing engineer’s listening environment influence the final experience of the home listener, especially regarding the role of reflections? A pilot study explores the effect of specular and diffuse lateral reflection on the perception of trained listeners. Reflections may strongly influence balance, equalization, dynamics, and reverberation. In contrast to earlier studies that used normal listeners, this study uses trained audio engineers to perform selected tasks in a variety of acoustic settings. A correlation was observed between the presence of strong lateral energy and an initial reduction of speed for performing a task. However, adaptation soon occurs, thereby restoring the subjects to normal accuracy and response time.
Separation of Harmonic Musical Instrument Notes Using Spectro-Temporal Modeling of Harmonic Magnitudes and Spectrogram Inversion with Phase Optimization
Separating the individual sources in a single channel is particularly difficult because of overlapping harmonics. Western music is often arranged so that sounds are not only occurring simultaneously, but are also harmonically related. While traditional approaches use either spectral or temporal models, the proposed model exploits the combination of the spectral and temporal correlations of harmonic magnitudes to estimate the regions of overlap. A diverse selection of harmonic musical instruments was analyzed, and a generalized-instrument magnitude track prediction model was derived to generate track estimates. This approach, which exploits dependencies among tracks, was shown to be consistently more accurate than existing methods.
Because of their high efficiency, low volume, and reduced weight, switched Class D audio amplifiers exist in a wide range of applications. However, there are difficult technical issues. This research presents a multiloop feedback topology for the PWM Class D high-quality audio amplifiers that includes: (a) the output filter inside the feedback loop, (b) the means for directly regulating the load voltage, and (c) compensation of the nonlinearities of the output filter. This approach guarantees a flat frequency response regardless of the load and provides a low THD even when smaller, less linear, and less expensive output filter components are used. The theory of the multiloop configuration is analyzed. The proposed topology was built into an integrated power amplifier, which demonstrates the effectiveness of this approach.
The parameters of noise-reduction algorithms in consumer products, such as hearing aids, are often preset using a generic model to represent the average listener. This research explores the degree to which tunable parameters improve the trade-off between noise reduction and speech degradation. A logistic probability model was developed that allowed for interpolation of the data so that the optimal trade-off for each individual subject could be determined. For five out of ten subjects, their preferred settings were modestly different from the parameters obtained using group averages. Nevertheless, at least for normal hearing listeners, individualization does not appear to be critical. Hearing-impaired listeners may acquire more benefit from custom tuning.
Although sound-field reproduction methods have been extensively studied over the past decades, there is no single approach that dominates. This paper describes a method called Co-Variance Method (CVM) that reproduces a spatial covariance matrix of the original sound field without knowledge of sound source locations. Conventional methods based on physical theory require an impractically large number of transducers. In contrast, CVM is based on the statistical properties of a sound field. Simulation experiments evaluated the relation between reproduction errors and the number of microphone-loudspeaker pairs when reconstructing the wave fronts of sinusoidal, impulse, and saw-tooth-like impulsive signals. With CVM, a wave front resembling that of the original field was obtained in the listening area.
Standards and Information Documents
AES Standards Committee News
Audio-file transfer and exchange; MPEG surround; AES3; connector for surround microphones; miniature XLR-type connectors; loudspeaker modeling and measurement; microphone measurement and characterization; audio connectors
133rd Convention Report, San Francisco
133rd Convention Exhibitors
The audio system in your car is now potentially more advanced than the one you have at home. People also spend longer listening in their cars than they do at home. This leads to a demand for higher levels of quality and sophistication, including better quality control, separate sound zones, and higher spatial complexity.
51st Conference, Helsinki, Call for Papers
Index to Volume 60