Sound Source Separation

This is the author's accepted pre-print of the article, first published as G. Evangelista, S. Marchand, M. D. Plumbley and E. Vincent. Sound source separation. In U. Zölzer (ed.), DAFX: Digital Audio Effects, 2nd edition, Chapter 14, pp. 551-588. John Wiley & Sons, March 2011. ISBN 9781119991298. DOI: 10.1002/9781119991298.ch14file: Proof:e\EvangelistaMarchandPlumbleyV11-sound.pdf:PDF owner: markp timestamp: 2011.04.26file: Proof:e\EvangelistaMarchandPlumbleyV11-sound.pdf:PDF owner: markp timestamp: 2011.04.2

Experimental study of acoustic displays of flight parameters in a simulated aerospace vehicle

Evaluating acoustic displays of target location in target detection and of flight parameters in simulated aerospace vehicle

Wireless recording of the calls of Rousettus aegyptiacus and their reproduction using electrostatic transducers

Bats are capable of imaging their surroundings in great detail using echolocation. To apply similar methods to human engineering systems requires the capability to measure and recreate the signals used, and to understand the processing applied to returning echoes. In this work, the emitted and reflected echolocation signals of Rousettus aegyptiacus are recorded while the bat is in flight, using a wireless sensor mounted on the bat. The sensor is designed to replicate the acoustic gain control which bats are known to use, applying a gain to returning echoes that is dependent on the incurred time delay. Employing this technique allows emitted and reflected echolocation calls, which have a wide dynamic range, to be recorded. The recorded echoes demonstrate the complexity of environment reconstruction using echolocation. The sensor is also used to make accurate recordings of the emitted calls, and these calls are recreated in the laboratory using custom-built wideband electrostatic transducers, allied with a spectral equalization technique. This technique is further demonstrated by recreating multi-harmonic bioinspired FM chirps. The ability to record and accurately synthesize echolocation calls enables the exploitation of biological signals in human engineering systems for sonar, materials characterization and imaging

KAVUAKA: a low-power application-specific processor architecture for digital hearing aids

The power consumption of digital hearing aids is very restricted due to their small physical size and the available hardware resources for signal processing are limited. However, there is a demand for more processing performance to make future hearing aids more useful and smarter. Future hearing aids should be able to detect, localize, and recognize target speakers in complex acoustic environments to further improve the speech intelligibility of the individual hearing aid user. Computationally intensive algorithms are required for this task. To maintain acceptable battery life, the hearing aid processing architecture must be highly optimized for extremely low-power consumption and high processing performance.The integration of application-specific instruction-set processors (ASIPs) into hearing aids enables a wide range of architectural customizations to meet the stringent power consumption and performance requirements. In this thesis, the application-specific hearing aid processor KAVUAKA is presented, which is customized and optimized with state-of-the-art hearing aid algorithms such as speaker localization, noise reduction, beamforming algorithms, and speech recognition. Specialized and application-specific instructions are designed and added to the baseline instruction set architecture (ISA). Among the major contributions are a multiply-accumulate (MAC) unit for real- and complex-valued numbers, architectures for power reduction during register accesses, co-processors and a low-latency audio interface. With the proposed MAC architecture, the KAVUAKA processor requires 16 % less cycles for the computation of a 128-point fast Fourier transform (FFT) compared to related programmable digital signal processors. The power consumption during register file accesses is decreased by 6 %to 17 % with isolation and by-pass techniques. The hardware-induced audio latency is 34 %lower compared to related audio interfaces for frame size of 64 samples.The final hearing aid system-on-chip (SoC) with four KAVUAKA processor cores and ten co-processors is integrated as an application-specific integrated circuit (ASIC) using a 40 nm low-power technology. The die size is 3.6 mm2. Each of the processors and co-processors contains individual customizations and hardware features with a varying datapath width between 24-bit to 64-bit. The core area of the 64-bit processor configuration is 0.134 mm2. The processors are organized in two clusters that share memory, an audio interface, co-processors and serial interfaces. The average power consumption at a clock speed of 10 MHz is 2.4 mW for SoC and 0.6 mW for the 64-bit processor.Case studies with four reference hearing aid algorithms are used to present and evaluate the proposed hardware architectures and optimizations. The program code for each processor and co-processor is generated and optimized with evolutionary algorithms for operation merging,instruction scheduling and register allocation. The KAVUAKA processor architecture is com-pared to related processor architectures in terms of processing performance, average power consumption, and silicon area requirements

The effects of asymmetric directional microphone fittings on acceptance of background noise

The present study investigated the effects of asymmetric directional microphone fittings (i.e., an omnidirectional microphone on one ear and a directional microphone on the other) on speech understanding in noise and acceptance of background noise in 15 full-time hearing aid users. Subjects were fitted binaurally with four directional microphone conditions (i.e., binaural omnidirectional, asymmetric right directional, asymmetric left directional and binaural directional microphones) using Siemens Intuis directional behind-the-ear hearing aids and comply earmolds. The results revealed that speech understanding in noise improved when using asymmetric directional microphones compared to binaural omnidirectional microphone fittings and were not significantly hindered compared to binaural directional microphone fittings. The results also revealed that listeners who wore asymmetric directional microphones were more likely to accept background noise (i.e., accept hearing aids) than listeners fitted with binaural omnidirectional microphones. Lastly, the results revealed that the ANLs were better for the binaural directional microphones when compared to the asymmetric directional microphones, maximizing listeners\u27 willingness to wear hearing aids in the presence of noise

Investigating the build-up of precedence effect using reflection masking

The auditory processing level involved in the build‐up of precedence [Freyman et al., J. Acoust. Soc. Am. 90, 874–884 (1991)] has been investigated here by employing reflection masked threshold (RMT) techniques. Given that RMT techniques are generally assumed to address lower levels of the auditory signal processing, such an approach represents a bottom‐up approach to the buildup of precedence. Three conditioner configurations measuring a possible buildup of reflection suppression were compared to the baseline RMT for four reflection delays ranging from 2.5–15 ms. No buildup of reflection suppression was observed for any of the conditioner configurations. Buildup of template (decrease in RMT for two of the conditioners), on the other hand, was found to be delay dependent. For five of six listeners, with reflection delay=2.5 and 15 ms, RMT decreased relative to the baseline. For 5‐ and 10‐ms delay, no change in threshold was observed. It is concluded that the low‐level auditory processing involved in RMT is not sufficient to realize a buildup of reflection suppression. This confirms suggestions that higher level processing is involved in PE buildup. The observed enhancement of reflection detection (RMT) may contribute to active suppression at higher processing levels

Space Time Exploration of Musical Instruments

Musical instruments are tools used to generate sounds for musical expression. Virtual Reality (VR) and Augmented Reality (AR) musical instruments create sounds that may be spatially disjointed from the instrument controls. Spatial audio processing can be used to position the Extended Reality (XR) musical instruments and their corresponding sounds in the same space. This dissertation investigates novel ways of combining spatial reverb models to improve the naturalness of XR musical instruments. Seven spatial reverb systems, combinations of a shoebox spatial reverb model, a raytracing spatial reverb model, and measured directional room impulse response convolution reverb, were compared in a pilot study. A novel hybrid system of synthetic early reflections and directional room impulse responses was preferred for naturalness when tested over headphones with three instruments created by the author: AR electric guitar, AR drumset, and VR Singing Kite. This research culminated in a concert, Spherical Sound Search, which showcased the preferred hybrid system, the three XR musical instruments, and four re-contextualized spatial audio effects (spatial looping, spatial delay, spatial feedback, and spatial compression). The three pieces in the concert explored different aspects of XR modalities and presented the novel system with spatial audio effects to a larger audience by rendering to an octophonic loudspeaker layout