Partially Matching Projection Decoding Method Evaluation Under Different Playback Conditions

The paper presents an unsupervised binaural scene analyzer that is capable of performing simultaneous operations, such as localization, detection and recognition of target speaker amidst in a reverberant and noise inferential environment. The proposed technique includes three main stages: sound source localization, speaker recognition and auditory saliency-based indexing system. The stage one involves the localization of target speaker by incorporating the binaural cues into Azimuth dependant GMM-EM classifier. During the second stage, the study proposes a Gabor-Hilbert Envelope Coefficient (GHEC) based spectro-temporal extractor as an efficient speaker recognition method which shows better robustness with minimum computational complexity. The Hilbert Envelope provides relevant acoustic information and also it improvises the performance of speaker identification process in different reverberant environments and SNR values. Later in the third stage, the auditory saliency based diarization is proposed as a process of indexing the speech contents based on the image identity of the target speaker. The proposed system may be used to catalogue the entire speech content with the corresponding image of the target speaker that finds wide range of applications, including teleconference, hands-free communication and meeting hall content localization and fast audio retrieval in a repository.

This paper presents an automatic sound source localization approach based on a combination of the basic time delay estimation sub-methods namely, Time Difference of Arrival (TDOA), and Steered Response Power (SRP) methods. The TDOA method is a fast but vulnerable approach for finding the sound source location in long distances and reverberant environments and is so sensitive in noisy situations. On the other hand, the conventional SRP method is time consuming, but a successful approach to accurately find sound source location in noisy and reverberant environments. Also, another SRP-based method, SRP Phase Transform (SRP-PHAT), has been suggested for the better noise robustness and more accuracy of sound source localization. In this paper, based on the combination of TDOA and SRP based methods, two approaches were proposed for sound source localization. In the first proposed approach called Classical TDOA-SRP, the TDOA method is used to find the approximate sound source direction and then SRP based methods were used to find the accurate location of sound source in the Field of View (FOV) which is obtained by the TDOA method. In the second proposed approach which called Optimal TDOA-SRP, for more reduction of computational processing time of SRP-based methods and better noise robustness, a new criterion has been proposed for finding the effective FOV which is obtained through the TDOA method. Experiments were carried out under different conditions confirming the validity of the purposed approaches.

The invention of the telephone about 150 years ago extended the range of verbal communication between humans beyond the bounds given by the power of their voices. Using this technology, however, was – and still is – inherently connected with some inconveniences. The talkers have to hold a handset such that the loudspeaker is close to their ear and the microphone is adjacent to their mouth. Even then, speech quality is reduced and ambient noise may be picked up. Replacing the handset by a microphone and a loudspeaker now positioned a – short – distance from the talker increases the loss of the transmission loop by say 20 dB [6]. Furthermore, the level of ambient noise collected by the microphone is increased and the echo from the loudspeaker signal is picked up. Methods of acoustic echo and noise control aimed to remedy these disadvantages exhibit a long history. Originally, efforts were focussed on the development of hands-free telephone systems. In the following section we will highlight some of the important steps toward today’s systems. We will also point out that the developments were always linked to the technology available at the time of their proposal. This introductory chapter will close with an overview of important current developments reported in the following chapters of this book in detail by international experts in the field of acoustic echo and noise control.

Sound source localization with less data is a challenging task. To address this problem, a novel sound source localization method based on compressive sensing theory is proposed in this paper. Specifically, a sparsity basis is first constructed for each microphone by shifting the audio signal recorded from one reference microphone. In this manner, the microphones except the reference one are allowed to capture audio signals under the sampling rate far below the Nyquist criterion. Next, the source positions are estimated by solving an $$l_1$$ minimization based on each frame of audio signals. Finally, a fine localization scheme is presented by fusing the estimated source positions from multiple frames. The proposed method can directly determine the number of sound sources in one step and successfully estimate the source positions in noisy and reverberant environments. Experimental results demonstrate the validity of the proposed method.

Low-frequency sound source localization in enclosed space based on time reversal method

Sound field reproduction has applications in music reproduction, spatial audio, sound environment reproduction, and experimental acoustics. Sound field reproduction can be used to artificially reproduce the spatial character of natural hearing. The objective is then to reproduce a sound field in a real reproduction environment. Wave field synthesis (WFS) is a known open-loop technology which assumes that the reproduction environment is anechoic. The room response thus reduces the quality of the physical sound field reproduction by WFS. In recent research papers, adaptive wave field synthesis (AWFS) was defined as a potential solution to compensate for these quality reductions from which WFS objective performance suffers. In this paper, AWFS is experimentally investigated as an active sound field reproduction system with a limited number of reproduction error sensors to compensate for the response of the listening environment. Two digital signal processing algorithms for AWFS are used for comparison purposes, one of which is based on independent radiation mode control. AWFS performed propagating sound field reproduction better than WFS in three tested reproduction spaces (hemianechoic chamber, standard laboratory space, and reverberation chamber).

One interesting aspect of sound-source localization in reverberant environments is that different stimuli (e.g., speech versus noise) can elicit different spatial sensations at the same delays between the direct sound and its reflections. For example, a given delay between a single simulated direct sound and a delayed copy presented from a different location may elicit the sensation of either a single sound located at or near its source for speech while a noise stimulus under otherwise the same conditions might be perceived as a direct sound with echoes. Furthermore, presenting stimuli over headphones versus loudspeakers can further modify spatial perception for the same stimulus conditions. This modeling study will explore reasons for these two interesting differences of behavioral outcome, using simulation tools developed in Montagne and Zhou (JASA, 2018) and Pastore and Braasch (JASA, 2020).

Introduction Researchers at The University of Melbourne are evaluating the broader outcomes of bilateral implantation in early implanted school-age children, including effects on parent and child stress (Sarant and Garrard, 2013), language (Sarant et al., 2014), psychosocial functioning, academic achievement, and everyday listening. The Speech, Spatial, and Other Qualities of Hearing Scale (SSQ) is a three-part adult questionnaire (http://www.ihr.mrc.ac.uk/products/ display/ssq; Gatehouse and Noble, 2004). The SSQ focusses mainly on hearing functions requiring the binaural system and is therefore suitable for evaluating everyday listening performance with bilateral implants. The SSQ-P is an adapted version for use with parents of children with hearing impairment (Galvin and Noble, 2013). The first section of the SSQ focusses on speech perception in different listening scenarios, such as in group conversation in a noisy environment, or a single speaker in a reverberant environment. The second section focusses on spatial hearing, in this case the direction of movement and location of sound sources, including the speaker in a group conversation. The final section focusses on other qualities of hearing, including sound segregation, listening effort, and sound and object identification.

Mobile robots communicating with people would benefit from being able to detect sound sources to help localize interesting events in real-life settings. We propose using a spherical robot with four microphones to determine the spatial locations of multiple sound sources in ordinary rooms. The arrival temporal disparities from phase difference histograms are used to calculate the time differences. A precedence effect model suppresses the influence of echoes in reverberant environments. To integrate spatial cues of different microphones, we map the correlation between different microphone pairs on a 3D map corresponding to the azimuth and elevation of sound source direction. Results of experiments indicate that our proposed system provides sound source distribution very clearly and precisely, even concurrently in reverberant environments with the Echo Avoidance (EA) model.

Article Acoustic noise control during auditory fMRI using a DSP system - first initial in vivo results was published on August 31, 2012 in the journal Biomedical Engineering / Biomedizinische Technik (volume 57, issue SI-1-Track-J).

Multiple simultaneous sound source localization (SSL) is one of the most important applications in the speech signal processing. The one-step algorithms with the advantage of low computational complexity (and low accuracy), and the two-step methods with high accuracy (and high computational complexity) are proposed for multiple SSL. In this article, a combination of one-step-based method based on the generalized eigenvalue decomposition (GEVD), and a two-step-based method based on the adaptive generalized cross-correlation (GCC) by using the phase transform/maximum likelihood (PHAT/ML) filters along with a novel T-shaped circular distributed microphone array (TCDMA) is proposed for 3D multiple simultaneous SSL. In addition, the low computational complexity advantage of the GCC algorithm is considered in combination with the high accuracy of the GEVD method by using the distributed microphone array to eliminate spatial aliasing and thus obtain more appropriate information. The proposed T-shaped circular distributed microphone array-based adaptive GEVD and GCC-PHAT/ML algorithms (TCDMA-AGGPM) is compared with hierarchical grid refinement (HiGRID), temporal extension of multiple response model of sparse Bayesian learning with spherical harmonic (SH) extension (SH-TMSBL), sound field morphological component analysis (SF-MCA), and time-frequency mixture weight Bayesian nonparametric acoustical holography beamforming (TF-MW-BNP-AHB) methods based on the mean absolute estimation error (MAEE) criteria in noisy and reverberant environments on simulated and real data. The superiority of the proposed method is presented by showing the high accuracy and low computational complexity for 3D multiple simultaneous SSL.

In designing a microphone array for speech acquisition in a reverberant room, one is often faced with a mixed nearfield/farfield design problem, i.e., design a beamformer which can focus on a nearfield source, but which simultaneously can cancel room reverberation (which is typically modeled as isotropic farfield interference). This paper presents a new technique to solve such a problem. Using the spherical solution to the wave equation, a beamforming technique is presented to simultaneously approximate a desired nearfield beampattern and a desired farfield beampattern.

Some kinds of thermofluid instruments are using Helmholtz resonators for absorbing noises radiated from them. Since the resonance frequency of a Helmholtz resonator depends on the ratio of the resonant volume to the neck size, operating frequency range can be easily tuned and thus the Helmholtz resonator is well-suited for noise control in a wide range of frequency. Besides the noise control application, Helmholtz resonators have been recently applied to acoustic energy harvesting. To date, most of studies on energy harvesting using Helmholtz resonators are employing piezoelectric materials located in the resonant volume and enhancing harvesting efficiency by tuning the resonance frequency and the structure of the added piezoelectric materials. In this study, a piezoelectric (PVDF composite) cantilever was integrated within a Helmholtz resonator. Then, energy harvesting efficiency was maximized by matching and tuning mechanical resonance of the piezoelectric cantilever and acoustic resonance of the Helmholtz resonator. Theoretical analysis including finite element simulation was also performed to confirm our experimental results. To maximize the harvest power, the electrical impedance matching was considered. This study can be applied to acoustic energy harvesting as well as acoustic noise control.

Rescue robots have been developed for search and rescue operations in times of large-scale disasters. Such a robot is used to search for survivors in disaster sites by capturing their voices with its microphone array. However, since the robot has many vibration motors, ego noise is mixed with voices, and it is difficult to differentiate the ego noise from a call for help from a disaster survivor. In our previous works, an ego noise reduction technique that combines a method of blind source separation called independent low-rank matrix analysis and postprocessing for noise cancellation was proposed. In the practical use of this robot, to determine the precise location of survivors, the direction of the observed voice should be estimated after the ego noise reduction process. To achieve this objective, in this study, a new hose-shaped rescue robot with microphone arrays was developed. Moreover, we adapt postfilter called MOSIE to our previous noise reduction method to listen to stereo sound because this robot can record stereo sound. By performing in a simulated disaster site, we confirm that the operator can perceive the direction of a survivor's location by applying a speech enhancement technique combining independent low-rank matrix analysis, noise cancellation, and postfiltering to the observed multichannel noisy signals.

One of the properties of composite materials is the possibility of having phononic band gaps, within which sound and vibrations at certain frequencies do not propagate. These materials are called Phononic Crystals (PCs). PCs with large band gaps are of great interest for many applications, such as transducers, elastic/ acoustic filters, noise control, and vibration shields. Most of previous works concentrates on PCs made of elastic isotropic materials; however, band gaps can be enlarged by using non-isotropic materials, such as piezoelectric materials. Since the main property of PCs is the presence of band gaps, one possible way to design structures which have a desired band gap is through Topology Optimization Method (TOM). TOM is a computational technique that determines the layout of a material such that a prescribed objective is maximized. Functionally Graded Materials (FGM) are composite materials whose properties vary gradually and continuously along a specific direction within the domain of the material. One of the advantages of applying the FGM concept to TOM is that it is not necessary a discrete 0-1 result, once the material gradation is part of the solution. Therefore, the interpretation step becomes easier and the dispersion diagram obtained from the optimization is not significantly modified. In this work, the main objective is to optimize the position and width of piezocomposite materials band gaps. Finite element analysis is implemented with Bloch-Floquet theory to solve the dynamic behavior of two-dimensional functionally graded unit cells. The results demonstrate that phononic band gaps can be designed by using this methodology.