The modeling of diffuse boundaries in the 2-D digital waveguide mesh

The digital waveguide mesh can be used to simulate the propagation of sound waves in an acoustic system. The accurate simulation of the acoustic characteristics of boundaries within such a system is an important part of the model. One significant property of an acoustic boundary is its diffusivity. Previous approaches to simulating diffuse boundaries in a digital waveguide mesh are effective but exhibit limitations and have not been analyzed in detail. An improved technique is presented here that simulates diffusion at boundaries and offers a high degree of control and consistency. This technique works by rotating wavefronts as they pass through a special diffusing layer adjacent to the boundary. The waves are rotated randomly according to a chosen probability function and the model is lossless. This diffusion model is analyzed in detail, and its diffusivity is quantified in the form of frequency dependent diffusion coefficients. The approach used to measuring boundary diffusion is described here in detail for the 2-D digital waveguide mesh and can readily be extended for the 3-D case

The KW-boundary hybrid digital waveguide mesh for room acoustics applications

The digital waveguide mesh is a discrete-time simulation used to model acoustic wave propagation through a bounded medium. It can be applied to the simulation of the acoustics of rooms through the generation of impulse responses suitable for auralization purposes. However, large-scale three-dimensional mesh structures are required for high quality results. These structures must therefore be efficient and also capable of flexible boundary implementation in terms of both geometrical layout and the possibility for improved mesh termination algorithms. The general one-dimensional N-port boundary termination is investigated, where N depends on the geometry of the modeled domain and the mesh topology used. The equivalence between physical variable Kirchoff-model, and scattering-based wave-model boundary formulations is proved. This leads to the KW-hybrid one-dimensional N-port boundary-node termination, which is shown to be equivalent to the Kirchoff- and wave-model cases. The KW-hybrid boundary-node is implemented as part of a new hybrid two-dimensional triangular digital waveguide mesh. This is shown to offer the possibility for large-scale, computationally efficient mesh structures for more complex shapes. It proves more accurate than a similar rectilinear mesh in terms of geometrical fit, and offers significant savings in processing time and memory use over a standard wave-based model. The new hybrid mesh also has the potential for improved real-world room boundary simulations through the inclusion of additional mixed modeling algorithms

Characteristics of the Terdiurnal tide in the MLT above Davis and Syowa

第3回極域科学シンポジウム　横断セッション「中層大気・熱圏」 11月26日（月） 国立極地研究所 ２階大会議

Real-time dynamic articulations in the 2-D waveguide mesh vocal tract model

Time domain articulatory vocal tract modeling in one-dimensional (1-D) is well established. Previous studies into two-dimensional (2-D) simulation of wave propagation in the vocal tract have shown it to present accurate static vowel synthesis. However, little has been done to demonstrate how such a model might accommodate the dynamic tract shape changes necessary in modeling speech. Two methods of applying the area function to the 2-D digital waveguide mesh vocal tract model are presented here. First, a method based on mapping the cross-sectional area onto the number of waveguides across the mesh, termed a widthwise mapping approach is detailed. Discontinuity problems associated with the dynamic manipulation of the model are highlighted. Second, a new method is examined that uses a static-shaped rectangular mesh with the area function translated into an impedance map which is then applied to each waveguide. Two approaches for constructing such a map are demonstrated; one using a linear impedance increase to model a constriction to the tract and another using a raised cosine function. Recommendations are made towards the use of the cosine method as it allows for a wider central propagational channel. It is also shown that this impedance mapping approach allows for stable dynamic shape changes and also permits a reduction in sampling frequency leading to real-time interaction with the model

Acoustic modeling using the digital waveguide mesh

The digital waveguide mesh has been an active area of music acoustics research for over ten years. Although founded in 1-D digital waveguide modeling, the principles on which it is based are not new to researchers grounded in numerical simulation, FDTD methods, electromagnetic simulation, etc. This article has attempted to provide a considerable review of how the DWM has been applied to acoustic modeling and sound synthesis problems, including new 2-D object synthesis and an overview of recent research activities in articulatory vocal tract modeling, RIR synthesis, and reverberation simulation. The extensive, although not by any means exhaustive, list of references indicates that though the DWM may have parallels in other disciplines, it still offers something new in the field of acoustic simulation and sound synth

Spectral Modelling Synthesis of Vehicle Pass-by Noise

Spectral Modelling Synthesis (SMS) is a sound synthesis technique that models time-varying spectra of given sounds as a collection of sinusoids plus a filtered noise component. Although originally utilized to produce musical sounds, this technique can also be extended for analysis, transformation and synthesis of a wide range of environmental sounds, such as traffic noise. Simplifications based on psychoacoustic analysis can be conducted during the modelling process to avoid redundant data, which leads to perceptual similarity between synthesized sounds and the original recordings of vehicle pass-by noise. In this paper, we investigate if this perceptual similarity can be described by objective metrics, and how to improve the synthesis by tuning the parameters in the SMS algorithm. The results showed that vehicle pass-by sounds characterized by tyre and engine noise can be well synthesized with different parameter sets in the SMS algorithm. Furthermore, it is found that Zwicker Roughness is a sensitive metric for measuring the perceptual similarity between original recordings and synthesized sounds as it varies significantly when tuning SMS parameters

The Antarctic radio tropopause, cirrus clouds and their relation to meteorological systems

第3回極域科学シンポジウム/第35回極域気水圏シンポジウム 11月30日（金） 国立国語研究所 ２階多目的

An evaluation of pre-processing techniques for virtual loudspeaker binaural ambisonic rendering

International audienceBinaural Ambisonic rendering is widely used in immersive applications such as virtual reality due to its sound field rotation capabilities. Binaural Ambisonic reproduction can theoretically replicate the original sound field exactly for frequencies up to what is commonly referred to as the `spatial aliasing frequency', f_alias. At frequencies above f_alias however, reproduction can become inaccurate due to the limited spatial accuracy of reproducing a physical sound field with a finite number of transducers, which in practice causes localisation blur, reduced lateralisation and comb filtering spectral artefacts. The standard approach to improving Ambisonic reproduction is to increase the order of Ambisonics, which allows for exact sound field reproduction up to a higher f_alias, though at the expense of more channels for storage, more microphone capsules for recording, and more convolutions in binaural reproduction. It is therefore highly desirable to explore alternative methods of improving low-order Ambisonic rendering. One common practice is to employ a dual-band decoder with basic Ambisonic decoding at low frequencies and Max r_E channel weighting above f_alias, which improves spectral, localisation and lateralisation reproduction. Virtual loudspeaker binaural Ambisonic decoders can be made by multiplying each loudspeaker's head-related impulse responses (HRIRs) with the decode matrix coefficients and summing the resulting spherical harmonic (SH) channels. This approach allows for dual-band decoding and loudspeaker configurations with more loudspeakers than SH channels whilst minimising the required number of convolutions. Binaural Ambisonic reproduction using the virtual loudspeaker approach is then achieved by a summation of direct convolution of each SH channel of the encoded signal with the corresponding SH channel of the binaural decoder. This paper presents the method and results of a perceptual comparison of state-of-the-art pre-processing techniques for virtual loudspeaker binaural Ambisonic rendering. By implementing these pre-processing techniques in the HRTFs used in the virtual loudspeaker binaural rendering stage, improvements can be made to the rendering. All pre-processing techniques are implemented offline, such that the resulting binaural decoders are of the same size and require the same number of real-time convolutions.The three pre-processing techniques investigated in this study are:\beginitemize ıtem Diffuse-field Equalisation (DFE) ıtem Ambisonic Interaural Level Difference Optimisation (AIO) ıtem Time Alignment (TA) \enditemizeDFE is the removal of direction-independent spectral artefacts in the Ambisonic diffuse-field. AIO augments the gains of the left and right virtual loudspeaker HRTF signals above f_alias such that Ambisonic renders produce more accurate interaural level differences (ILDs). TA is the removal of interaural time differences (ITDs) between the HRTFs above f_alias to reduce high frequency comb filtering effects.The test follows the multiple stimulus with hidden reference and anchors (MUSHRA) paradigm, ITU-R BS.1534-3. Tests are conducted in a quiet listening room using a single set of Sennheiser HD~650 circum-aural headphones and an Apple Macbook Pro with a Fireface UCX audio interface, which has software controlled input and output levels. Headphones are equalised from the RMS average of 11 impulse response measurements, with 1 octave band smoothing in the inverse filter. All audio is 24-bit depth and 48~kHz sample rate. Listening tests are conducted using first, third and fifth order Ambisonics, with respective loudspeaker configurations comprising 6, 26 and 50 loudspeakers, arranged in Lebedev grids. The different test conditions are made up of various combinations of the three pre-processing techniques. The test conditions are as follows:\beginenumerate ıtem HRTF convolution (reference) ıtem Standard Ambisonic (dual band) ıtem Ambisonic with DFE (dual band) ıtem Ambisonic with AIO (dual band) ıtem Ambisonic with AIO & DFE (dual band) ıtem Ambisonic with TA & DFE (basic) ıtem Ambisonic with TA & AIO & DFE (basic) ıtem Ambisonic with TA & AIO & DFE (dual band)\endenumerateThe stimuli are synthesised complex acoustic scenes, defined in this paper as an acoustic scene with multiple sources. The synthesised complex scene used in this paper is composed from 24 freely available stems of an orchestra. Instruments are isolated and empirically matched in loudness. The orchestral stems are panned to the vertices of a 24 pt. T-design arrangement, to ensure minimal overlap between virtual loudspeaker positions in the binaural decoders and the sound sources in the complex scene. Synthesising complex scenes in this way allows for an explicit target reference stimulus - in this case a direct HRTF convolved render. If the Ambisonic stimuli are perfectly reconstructed, they will be equivalent to the reference stimulus. Results are analysed using non-parametric statistics and discussed in the full manuscript. The conclusion suggests the perceptually preferred pre-processing algorithms for virtual loudspeaker binaural Ambisonic rendering.</latex&gt

Silent speech: restoring the power of speech to people whose larynx has been removed

Every year, some 17,500 people in Europe and North America lose the power of speech after undergoing a laryngectomy, normally as a treatment for throat cancer. Several research groups have recently demonstrated that it is possible to restore speech to these people by using machine learning to learn the transformation from articulator movement to sound. In our project articulator movement is captured by a technique developed by our collaborators at Hull University called Permanent Magnet Articulography (PMA), which senses the changes of magnetic field caused by movements of small magnets attached to the lips and tongue. This solution, however, requires synchronous PMA-and-audio recordings for learning the transformation and, hence, it cannot be applied to people who have already lost their voice. Here we propose to investigate a variant of this technique in which the PMA data are used to drive an articulatory synthesiser, which generates speech acoustics by simulating the airflow through a computational model of the vocal tract. The project goals, participants, current status, and achievements of the project are discussed below.Universidad de Málaga. Campus de Excelencia Internacional Andalucía Tech

Analysis and Resynthesis of the Handpan Sound

(Abstract to follow