Data-driven Threshold Selection for Direct Path Dominance Test

Direction-of-arrival estimation methods, when used with recordings made in enclosures are negatively affected by the reflections and reverberation in that enclosure. Direct path dominance (DPD) test was proposed as a pre-processing stage which can provide better DOA estimates by selecting only the time-frequency bins with a single dominant sound source component prior to DOA estimation, thereby reducing the total computational cost. DPD test involves selecting bins for which the ratio of the two largest singular values of the local spatial correlation matrix is above a threshold. The selection of this threshold is typically carried out in an ad hoc manner, which hinders the generalisation of this approach. This selection method also potentially increases the total computational cost or reduces the accuracy of DOA estimation. We propose a DPD test threshold selection method based on a data-driven statistical model. The model is based on the approximation of the singular value ratio distribution of the spatial correlation matrices as a generalised Pareto distribution and allows selecting time-frequency bins based on their probability of occurrence. We demonstrate the application of this threshold selection method via emulations using acoustic impulse responses measured in a highly reverberant room with a rigid spherical microphone array

Scattering Delay Network Simulator of Coupled Volume Acoustics

IEEEArtificial reverberators provide a computationally viable alternative to full-scale room acoustics simulation methods for deployment in interactive, immersive systems. Scattering delay network (SDN) is an artificial reverberator that allows direct parametric control over the geometry of a simulated cuboid enclosure as well as the directional characteristics of the simulated sound sources and microphones. This paper extends the concept of SDN reverberators to multiple enclosures coupled via an aperture. The extension allows independent control of the acoustical properties of the coupled enclosures and the size of the connecting aperture. The transfer function of the coupled-volume SDN system is derived. The effectiveness of the proposed method is evaluated in terms of rendered energy decay curves in comparison to full-scale ray-tracing models and scale model measurements

Source excitation strategies for obtaining impulse responses in finite difference time domain room acoustics simulation

This paper considers source excitation strategies in finite difference time domain room acoustics simulations for auralization purposes. We demonstrate that FDTD simulations can be conducted to obtain impulse responses based on unit impulse excitation, this being the shortest, simplest and most efficiently implemented signal that might be applied. Single, rather than double, precision accuracy simulations might be implemented where memory use is critical but the consequence is a remarkably increased noise floor. Hard source excitation introduces a discontinuity in the simulated acoustic field resulting in a shift of resonant modes from expected values. Additive sources do not introduce such discontinuities, but instead result in a broadband offset across the frequency spectrum. Transparent sources address both of these issues and with unit impulse excitation the calculation of the compensation filters required to implement transparency is also simplified. However, both transparent and additive source excitation demonstrate solution growth problems for a bounded space. Any of these approaches might be used if the consequences are understood and compensated for, however, for room acoustics simulation the hard source is the least favourable due to the fundamental changes it imparts on the underlying geometry. These methods are further tested through the implementation of a directional sound source based on multiple omnidirectional point sources

Analiz-temelli Sentez Yöntemleriyle Uzamsal Ses Üretimi

TÜBİTAK EEEAG Proje01.03.2018Bu projenin amacı terminalden bağımsız, sondan-sona bir nesne-temelli ses üretimi yöntemi geliştirilmesidir. Bu amaca yönelik olarak 1) açık küresel mikrofon dizisi tasarımı ve gerçekleştirilmesi, 2) ses nesnelerinin kaydedilmesine olanak sağlayacak mikrofon dizisi sinyal işleme yöntemleri geliştirilmesi, 3) ses sahnelerinin betimlenmesine olanak sağlayan bir metadata biçemi geliştirilmesi, 4) ses sahnesinin düzenlenebilmesine olanak sağlayacak bir editör geliştirilmesi ve 5) ses sahnelerinin etkileşimli olarak geri çatılabilmesini sağlayacak esnek yöntemler geliştirilmesi planlanmıştır. Projenin ilk çıktılarından biri 13 mikrofondan oluşan ve akustik yeğinlik ölçümüne olanak sağlayan bir açık küresel mikrofon dizisinin tasarımı ve uygulanması olmuştur. Bu mikrofon dizisinin kalibrasyonu ve testleri yapılmış ve bir sonraki adımda geliştirilen bazı algoritmalarda kullanılacak olan dürtü cevabı ölçümlerinin yapılmasında kullanılmıştır. Mikrofon dizisi sinyal işleme alanında yapılan çalışmalarda ses varış yönü kestirimi ve ses kaynak ayırma işlemlerinde kullanılacak açık ve kapalı mikrofon dizileri için ayrı ayrı olmak üzere yeni ve özgün yöntemler geliştirilmiştir. Açık küresel mikrofon dizileri için geliştirilen yöntemler, akustik yeğinlik temelli varış yönü kestirimi algoritmaları ve dördey sinyal işleme temelli ses kaynak ayırma yöntemleri olmuştur. Kapalı küresel mikrofon dizileri için ise küresel harmonik alanda uzamsal entropi kavramını kullanan yeni bir varış yönü kestirimi yöntemi ve karmaşık dikgen eşleştirmeli izleme yöntemini kullanan bir ses kaynak ayırma yöntemi geliştirilmiştir. Ses sahnelerinin betimlenmesine olanak sağlayacak, SpatDIF biçemini genişleten yeni bir metaveri biçemi tasarlanmış ve bu biçemi düzenlemeye olanak sağlayan görsel bir editör tasarlanmıştır. Son olarak, ses sahnelerinin geri çatılmasında kullanılmak üzere gerçek zamanda çalışabilen bir oda akustiği simülatörü / yapay yankışımcı geliştirilmiştir. Bu simülatörün gerçekçiliğini arttırmak için birbirine bağlı hacimler ve kırılım modellerinin sistemle tümleştirilmesi çalışmaları yapılmış ve başarılı sonuçlar alınmıştır. Proje sonucunda iki dergi ve iki konferans yayını yapılmıştır. Bu yayınlara ek olarak Mart 2018?de bir dergi makalesi, Nisan 2018?de ise bir yeni konferans bildirisi değerlendirilmek üzere gönderilmiştir. Ayrıca biri yurtdışında davetli konuşma olmak üzere iki eğitim semineri verilmiştir.The aim of this project is to develop a terminal-agnostic, end-to-end object-based audio reproduction system. To that aim, the work carried out in the project consisted of 1) the design and development of an open spherical microphone array, 2) development of microphone array signal processing methods, 3) development of a scene description metadata format, 4) design and development of a sound scene editor, and 5) development of flexible synthesis methods that can be used to reconstruct the intended sound field. One of the first outcomes of the project is a 13-channel open spherical microphone array that allows the measurement of acoustic intensity. This array was calibrated, tested and employed in recording acoustic impulse responses to be used later on with the algorithms that were developed in the subsequent stages of the project. In the field of microphone array signal processing, the emphasis was on the development of novel algorithms for direction-of-arrival (DOA) estimation and acoustic source separation both for open and rigid spherical microphone arrays. Methods developed for open spherical microphone arrays used acoustic intensity as a basis for DOA estimation and quaternion signal processing methods for source separation. Methods developed for rigid spherical microphone arrays operate in the spherical harmonic domain and use spatial entropy for DOA estimation and complex orthogonal matching pursuits (OMP) for acoustic source separation. A new metadata format for sound scene description which augments SpatDIF was developed alongside a visual tool which allows editing the metadata. Finally, a room acoustics simulator / artificial reverberator was designed to allow interactively reconstructing sound scenes. In order to extend the capabilities of this simulator, models of coupled volumes and edge diffraction were integrated with SDN-type reverberators with good results. Two journal articles and two conference papers were published during the project. In addition, a journal article was submitted in March 2018 and a conference papers was submitted in April 2018. Two lectures, one of which was an invited lecture abroad, based on the project findings were given

Procedural Synthesis of Gunshot Sounds Based on Physically Motivated Models

Generation of content for games is one of the major bottlenecks in terms of the effort required and the resources to be committed. A typical AAA game contains tens of thousands of sound files as audio assets which include spoken dialogue as well as sound effects. Procedural content generation (PCG) provides a cost effective alternative to recording these sounds in the studio or in the field. While some sound effects can be recorded fairly easily given the necessary time, effort, and resources, some others such as gunshot sounds are not easy to record. Since many games and simulations incorporate firearms, parametric sound synthesis which is essentially a PCG technique can be used to alleviate the need to record gunshot sounds. This chapter describes a physically-motivated parametric gunshot sound synthesis model. The model is based on a deconstruction of the gunshot sound event into its constituent parts and uses parameters such as the barrel length, bullet type, and muzzle velocity to synthesise the sounds of different firearms. A subjective evaluation which investigates the perceptual relevance of the proposed model is also presented

Modelling sound source localization under precedence effect using multivariate Gaussian mixtures

The precedence effect refers to the property of the human auditory system that enables accurate localization of sound sources where many interfering echoes of the original signal are also present. Perception of the elevation, azimuth, and distance of sound sources are affected in the presence of an echo. The multivariate Gaussian mixture model proposed in this paper combines azimuth, elevation and distance perception, and provides a general framework for modeling sound source localization under the precedence effect. The model interprets the precedence effect as a spatial property rather than a temporal one