144th AES CONVENTION Paper Session P23: Audio Processing and Effects — Part 2

AES Milan 2018
Paper Session P23

P23 - Audio Processing and Effects – Part 2

Saturday, May 26, 09:00 — 11:30

Balázs Bank, Budapest University of Technology and Economics - Budapest, Hungary

P23-1 Stage Compression in Transaural AudioFilippo Maria Fazi, University of Southampton - Southampton, Hampshire, UK; Eric Hamdan, University of Southampton - Southampton, UK
The reproduction of binaural audio with loudspeakers, also referred to as transaural audio, is affected by a number of artifacts. This work focuses on the effect of reproduction error on low frequency Interaural Time Difference (ITD). Transaural systems do not provide perfect cross-talk cancellation between the left and right ear signals, especially at low frequencies. It is shown that increase in cross-talk leads to a perceived source azimuth angle that is smaller than intended. The authors show that in ideal theoretical conditions the angular error calculated from the interaural phase difference indicates stage compression for frequencies for which high cross-talk occurs. This trend is shown in the resultant ITD calculated from Interaural Cross Correlation (IACC), examined in one-third octave bands.
Convention Paper 10012

P23-2 Multi-Track Crosstalk Reduction Using Spectral SubtractionFabian Seipel, TU Berlin - Berlin, Germany; Alexander Lerch, Georgia Institute of Technology - Atlanta, GA, USA
While many music-related blind source separation methods focus on mono or stereo material, the detection and reduction of crosstalk in multi-track recordings is less researched. Crosstalk or “bleed” of one recorded channel in another is a very common phenomenon in specific genres such as jazz and classical, where all instrumentalists are recorded simultaneously. We present an efficient algorithm that estimates the crosstalk amount in the spectral domain and applies spectral subtraction to remove it. Randomly generated artificial mixtures from various anechoic orchestral source material were employed to develop and evaluate the algorithm, which scores an average SIR-Gain result of 15.14 dB on various datasets with different amounts of simulated crosstalk.
Convention Paper 10013

P23-3 Wave Digital Modeling of the Diode-Based Ring ModulatorAlberto Bernardini, Politecnico di Milano - Milan, Italy; Kurt James Werner, Queen's University Belfast - Belfast, UK; Sonic Arts Research Centre (SARC); Paolo Maffezzoni, Politecnico di Milano - Milan, Italy; Augusto Sarti, Politecnico di Milano - Milan, Italy
The ring modulator is a strongly nonlinear circuit common in audio gear, especially as part of electronic musical instruments. In this paper an accurate model based onWave Digital (WD) principles is developed for implementing the ring modulator as a digital audio effect. The reference circuit is constituted of four diodes and two multi-winding transformers. The proposed WD implementation is based on the Scattering Iterative Method (SIM), recently developed for the static analysis of large nonlinear photovoltaic arrays. In this paper SIM is shown to be suitable for implementing also audio circuits for Virtual Analog applications, such as the ring modulator, since it is stable, robust and comparable to or more efficient than state-of-the-art strategies in terms of computational cost.
Convention Paper 10015

P23-4 Improving the Frequency Response Magnitude and Phase of Analogue-Matched Digital FiltersJohn Flynn, Balance Mastering - London, UK; Queen Mary University of London - London, UK; Joshua D. Reiss, Queen Mary University of London - London, UK
Current closed-form IIR methods for approximating an analogue prototype filter in the discrete-domain do not match frequency response phase. The frequency sampling method can match phase, but requires extremely long filter lengths (and corresponding latency) to perform well at low frequencies. We propose a method for discretizing an analogue prototype that does not succumb to these issues. Contrary to the IIR methods, it accurately approximates the phase, as well as the magnitude response. The proposed method exhibits good low frequency resolution using much smaller filter lengths than design by frequency sampling.
Convention Paper 10016

P23-5 Optimization of Personal Audio Systems for Intelligibility ContrastDaniel Wallace, University of Southampton - Southampton, UK; Jordan Cheer, University of Southampton - Southampton, Hampshire, UK
Personal audio systems are designed to deliver spatially separated regions of audio to individual listeners. This paper demonstrates a method of personal audio system design that provides a level of contrast in the perceived speech intelligibility between bright and dark audio zones. Limitations in array directivity which would lead to a loss of privacy are overcome by reproducing a synthetic masking signal in the dark zone. This signal is optimized to provide effective masking whilst remaining subjectively pleasant to listeners. Results of this optimization from a simulated personal audio system are presented.
Convention Paper 10017

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