Directional Impulse Responses Research Articles

A study on constructing arbitrary directivity of sound sources by fitting the spherical harmonics coefficients

We attempt to obtain room impulse responses by sound sources with arbitrary directional characteristics by directivity synthesis in the spherical harmonic domain. For this purpose, we use directional room impulse responses measured with a general loudspeaker pointed in multiple directions. Specifically, we attempt to synthesize the directivity from measurements in 24 directions, including the horizontal and elevation angles of 45 degrees each and the case where the loudspeaker was pointed directly above and below. This enables us to construct an omnidirectional sound source that strictly meets ISO standards or a more flexible directional that reflects musical instruments' directivity. For the omnidirectional case, we are also considering how much we can reduce the number of measurement directions and propose a practical index. In addition, because the proposed method uses a commercially available coaxial loudspeaker with enough sound quality and performs frequency spectrum correction at the same time as directional synthesis, it is possible to synthesize room impulse responses that can be used for auralization.

Application of sound field reproduction system to acoustic welfare engineering - Real time sound field transmission system for parent and child room in a concert hall -

This study attempted to implement a real-time sound field reproduction system for a parent-child (PC) room in a concert hall as an example of acoustic welfare engineering. The PC room is typically located at the rear of a hall where visitors with their children can enjoy performances without hesitation, but the sound quality is rarely discussed. This study attempts to improve this situation using a multi-channel reproduction system. Specifically, we implemented a system in which the directional impulse response of the hall measured by a microphone array is convolved in real-time with the sound played and captured separately on stage and reproduced by a loudspeaker array installed in a PC room. The 24-channel system, in which the polar and azimuth angles are both 45 degrees apart, our standard procedure, was found to have a high degree of physical reproducibility. Still, it needed to be more practical considering poor room conditions such as large windows. Therefore, we introduced panning using loudspeakers above and below the window frame instead of using the front loudspeakers. The impulse responses of the original and reproduced sound fields were compared, and it was found that the reproducibility was maintained even with the introduction of the panning.

Time reversal acoustic focusing in a reverberant room: Elongating a focus, and the effect of head-scattering

Time reversal acoustics (TRA) has tremendous potential for focusing, or sound concentration, in reverberant architectural environments. While TRA provides an inherently simple means of achieving a high degree of spatiotemporal focusing, its appropriateness for listening applications in the built environment is challenged by the need to accommodate a person within a sufficiently large focal area. At its simplest, a reciprocal time reversal focusing system for audible range sound will produce a focus with a spatial spread corresponding to one-eighth of the wavelength, which is inconveniently small for speech frequencies. However, an elongated focus is introduced by deploying a directional impulse response, which can be created with a directional microphone, or synthesized by time-delay-mixing of multiple omnidirectional impulse responses. This paper reports on results of physical experiments conducted in a reverberant room, complemented by finite-different time-domain simulations in which the effect of introducing a head-sized scatterer was investigated.

МЕТОД ГЕНЕРАЦИИ СВЕРХШИРОКОПОЛОСНЫХ СВЧ-СИГНАЛОВ С ЛИНЕЙНО-ЧАСТОТНОЙ МОДУЛЯЦИЕЙ НА ОСНОВЕ САМОГЕТЕРОДИНИРОВАНИЯ ИЗЛУЧЕНИЯ ЛАЗЕРНОГО ДИОДА

In this paper, microwave photonic technique of ultrawideband linearly frequency modulated signals generation is proposed. The proposed method based on direct modulation of the laser diode current, which provides a periodic change in the emission frequency. The Michelson interferometer summarizes laser signal and its replica with time delay exceeding 10 ms, which, in turn, is defined by time difference between arms. Optoelectronic conversion is performed in high-speed high-power photodiode. The proposed approach makes it possible to generate microwave signals with linear frequency modulation in the frequency range up to several tens of gigahertz by means of laser pump current modulation in the frequency range less than 100 MHz. The method of ultra-wideband microwave signals generation is implemented using a Michelson fiber-optic interferometer with Faraday mirrors. Such interferometer topology provides a doubling of the path length, and reduced gyroscopic effects due to a counter-propagating double pass. Changing the parameters of the generated microwave ultrawideband signals with frequency modulation is achieved by pump current shape and amplitude control. It is also proposed to use real-time oscilloscope without need of changing of the internal structure of such generator to pre-tune it. Time-frequency analysis is used to extract frequency versus time dependence. By so doing, direct impulse response estimation is performed. This method allows to characterize laser diode frequency response for arbitrary microwave waveform generation by means of pump current waveform calculation from experimental data.

Single and Multiple Continuous-Wave Interference Suppression Using Adaptive IIR Notch Filters Based on Direct-Form Structure in a QPSK Communication System

The removal filter coefficients in this technique are dependent on the jammer’s power and its Instantaneous Frequency (IF) information, which can both be obtained in the time–frequency domain (adaptive filtering techniques). The dependence of the removing/reducing filter characteristics on the interference power is critical, as it allows an optimal trade-off between removal interference and the amount of self-noise generated by the filter. This trade-off is bounded by the two extreme cases of no notch filter (no self-noise) and full suppression (k1 = 1) for both low- and high-power jammer values. In this paper, a cascade second-order adaptive direct Infinite Impulse Response (IIR) Notch Filter (NF) with a gradient-based algorithm to suppress the Continuous-Wave (CW and MCW) interference is proposed for maximizing the receiver Signal-to-Noise Ratio (SNR) in a Quadrature Phase-Shift Keying (QPSK)-modulated signal. The suppression approach consists of two Adaptive IIR NFs (ANFs) based on a direct-form structure: the Hd1(z) and Hd1(z). The proposal in this work presents a low-complexity Time-Domain (TD) algorithm for controlling the update filter coefficient and notch depth. Simulation results demonstrate that the proposed approach represents an effective method for removing/reducing the impacts of CWI/MCWI, resulting in improved system performance for low- and high-power jammer values when compared with the case of full suppression (k1 = 1); furthermore, it also improves the notch filter’s output SNR for a given Jamming-to-Signal Ratio (JSR) value and Bit Error Ratio (BER) performance. For example, the SNR output of the proposed IIR NF was enhanced by 7 dB versus the case without a filter when Eb/No = 15 dB and JSR = −5 dB. The proposed method can detect and mitigate weak and strong jamming with JSR values ranging from −30 to 40 dB, and can track the hopping frequency interference. Moreover, an improved BER performance is seen as compared to the case without an IIR NF.

Interpolating the Directional Room Impulse Response for Dynamic Spatial Audio Reproduction

Virtual reality (VR) is increasingly important for exploring the real world, which has partially moved to virtual workplaces. In order to create immersive presence in a simulated scene for humans, VR needs to reproduce spatial audio that describes three-dimensional acoustic characteristics in the counterpart physical environment. When the user moves, this reproduction should be dynamically updated, which provides practical challenges because the bandwidth for continuously transmitting audio and video scene data may be limited. This paper proposes an interpolation approach for dynamic spatial audio reproduction using acoustic characteristics of direction and reverberation at limited numbers of positions, which are represented using a first order Ambisonics encoding of the room impulse response (RIR), called the directional RIR (DRIR). We decompose two known DRIRs into reflection components, before interpolating early dominant components for DRIR synthesis and utilizing DRIR recordings for accuracy evaluation. Results indicate that the most accurate interpolation is obtained by the proposed method over two comparative approaches, particularly in a simulated small room where most direction of arrival estimation errors of early components are below five degrees. These findings suggest precise interpolated DRIRs with limited data using the proposed approach, which is vital for dynamic spatial audio reproduction for VR applications.

General method to mitigate the continuous wave interference and narrowband interference for GNSS receivers

A low-complexity method based on the second-order direct infinite impulse response (IIR) notch and its cascaded form for global navigation satellite systems (GNSSs) is proposed to mitigate the continuous wave interference (CWI) and narrowband interference (NI) as well as their mixed interference. The number of the cascaded stages is selected by the bandwidth of NI as well as that of the second-order direct IIR notch. The parameters of the second-order direct IIR notch and the bandwidth of NI are determined by the fast Fourier transform. To evaluate the performance of the proposed method, the authors conduct the simulation experiment and real-data experiment. In each experiment, the CWI scenario, the NI scenario and the mixed interference of CWI and NI scenarios are verified. The results of experiments depict the effectiveness of the introduced method.

Denoising Directional Room Impulse Responses with Spatially Anisotropic Late Reverberation Tails

Directional room impulse responses (DRIR) measured with spherical microphone arrays (SMA) enable the reproduction of room reverberation effects on three-dimensional surround-sound systems (e.g., Higher-Order Ambisonics) through multichannel convolution. However, such measurements inevitably contain a nondecaying noise floor that may produce an audible “infinite reverberation effect” upon convolution. If the late reverberation tail can be considered a diffuse field before reaching the noise floor, the latter may be removed and replaced with an extension of the exponentially-decaying tail synthesized as a zero-mean Gaussian noise. This has previously been shown to preserve the diffuse-field properties of the late reverberation tail when performed in the spherical harmonic domain (SHD). In this paper, we show that in the case of highly anisotropic yet incoherent late fields, the spatial symmetry of the spherical harmonics is not conducive to preserving the energy distribution of the reverberation tail. To remedy this, we propose denoising in an optimized spatial domain obtained by plane-wave decomposition (PWD), and demonstrate that this method equally preserves the incoherence of the late reverberation field.

Parametric inverse impulse response based on reduced order modeling and randomized excitations

This paper is concerned with the computation of the inverse impulse response of a parametrized structural dynamics problem using reduced-order modeling and randomized excitations. A two-stages approach is proposed, involving the solution of both direct and inverse problems. In the first stage, the parametrized structural dynamics problem is formulated in the frequency domain, and solved using a reduced-order modeling approach. As a result, the parametric transfer function of the structure is obtained, and then readily transformed into a parametric direct impulse response (DIR). In the second stage, the parametric inverse impulse response (IIR) is computed. We use randomized excitations to generate synthetic samples inexpensively from the parametric DIR. Based on these, the parametric IIR is computed by minimizing the mean square error between the estimate and the samples. Most importantly, we show that the randomized excitations can be generated by sampling the frequency domain only. Hence, the parametric domain does not need to be sampled, which makes the computation of the parametric IIR very efficient.

Performance comparison of various strategies in 24-channel versatile sound field reproduction system

A sound field reproducing system which consists of a hedgehog-shaped 24-channel narrow directional microphone and the speakers of the same number is currently proposed. The system is aimed for the versatile usage, for example, the reproduction of the sound field in a concert hall, and also applicable to various simulation for noise control engineering. The basic reproduction technique is intuitive and straightforward, in which a signal recorded by a particular microphone would be emitted from a loudspeaker located at the almost same direction. To compensate for the insufficient directional characteristics in the low-frequency range, some kinds of signal processing are necessary. So far, several methods, such as the inverse filtering based on the boundary surface control principle, the ambisonics, the beamforming, are attempted. The processed low-frequency components are integrated with the directly assigned high-frequency components. In addition to the procedure that uses 24 recorded signals, the alternative method which reproduces the directional information by convolving the measured directional impulse responses with the signal of the fewer channel, such as monaural or stereo format, could be used. The performance of these various reproduction methods is compared with each other.A sound field reproducing system which consists of a hedgehog-shaped 24-channel narrow directional microphone and the speakers of the same number is currently proposed. The system is aimed for the versatile usage, for example, the reproduction of the sound field in a concert hall, and also applicable to various simulation for noise control engineering. The basic reproduction technique is intuitive and straightforward, in which a signal recorded by a particular microphone would be emitted from a loudspeaker located at the almost same direction. To compensate for the insufficient directional characteristics in the low-frequency range, some kinds of signal processing are necessary. So far, several methods, such as the inverse filtering based on the boundary surface control principle, the ambisonics, the beamforming, are attempted. The processed low-frequency components are integrated with the directly assigned high-frequency components. In addition to the procedure that uses 24 recorded signals, the alternat...

Analysis of sound field isotropy based on directional energy decay curves

The analysis of the diffuseness of sound fields is of great interest in room acoustic applications ranging from the analysis of concert venues to reverberation room design and calibration. However, a standardized robust diffuseness estimation method is currently lacking. The fundamental definition of the diffuse sound field is that it is isotropic—requiring the sound field to be composed of infinitely many sound waves from uncorrelated sources with directions of arrival uniformly distributed over a sphere. Due to their symmetry, spherical microphone arrays are especially favorable for the analysis of the isotropy requirement. In this work, we propose the directional energy decay curve which we calculate from a directional room impulse response captured with a spherical microphone array. Further, we show how the analysis of the spatial variations of the directional energy decay curve enables examining the isotropy of sound fields in rooms. Finally, we present a simulation study of multiple room configurations with varying degrees of sound field diffuseness.

Scene-aware audio for 360° videos

Although 360° cameras ease the capture of panoramic footage, it remains challenging to add realistic 360° audio that blends into the captured scene and is synchronized with the camera motion. We present a method for adding scene-aware spatial audio to 360° videos in typical indoor scenes, using only a conventional mono-channel microphone and a speaker. We observe that the late reverberation of a room's impulse response is usually diffuse spatially and directionally. Exploiting this fact, we propose a method that synthesizes the directional impulse response between any source and listening locations by combining a synthesized early reverberation part and a measured late reverberation tail. The early reverberation is simulated using a geometric acoustic simulation and then enhanced using a frequency modulation method to capture room resonances. The late reverberation is extracted from a recorded impulse response, with a carefully chosen time duration that separates out the late reverberation from the early reverberation. In our validations, we show that our synthesized spatial audio matches closely with recordings using ambisonic microphones. Lastly, we demonstrate the strength of our method in several applications.

Room reflections analysis with the use of spherical beamforming and wavelets

Beamforming algorithms are widely used nowadays to localize sound sources in several types of environments. In general, the goal is to identify the direction from where a given signal is arriving. It consists of a spatial analysis performed over 2D maps, according to the frequency bands of interest or to a given spectral characteristic. In this paper, a spherical microphone array structure is developed. The signals obtained are input to Wavelet transform, in order to decompose impulse responses for real rooms. The objective is to extend the beamforming array capacity to identify and extract energy peaks from measured room impulse responses. The detected peaks represent sound reflections over the room surfaces, which can be used to describe acoustically the environment studied. With the proposed structure, it is possible to create acoustic maps, discretized in time and frequency, extending the applicability of traditional beamforming to provide directional impulse responses in reverberant spaces. The beamforming combines spherical harmonic decomposition with the Delay-and-Sum algorithm, whose result is post-processed through an Wavelet Transform. The system is applied to the array impulse response and allows to identify the incidence time and direction of the early reflections. The results are compared with those provided by ray-tracing simulation. A very good agreement is obtained for the detection of the first order reflections, besides the physical and spatial constraints of the spherical array developed.

Macroeconomic Determinants of Foreign Direct Investment Inflows and Impulse Response Function

The purpose of the contemporary research was the examination of macroeconomic determinants of foreign direct investment inflows into post-conflict Sierra Leone for the period, 2002-2015, and whether the determinants have a long-run and short-run association with FDI. A complementary exploration was the likely impulse response effect on FDI and GDP of a shock to FDI. The research method was quantitative, applying time series data with 56 observations and 6 variables. Applying the Phillips-Ouliaris cointegration model, the results indicated a statistically significant long-run relationship between FDI inflows and its determinants (Rho = -17.064, Tau = -2.996, p

DOA (direction of arrival) estimation of incident sound waves on a spherical microphone array: Comparison of some correction methods proposed to solve the DOA bias

A spherical 16-microphones array has been designed and built for room acoustics applications (measurement of directional room impulse responses or DRIR). The identification of strong specular reflections in a DRIR requires an accurate determination of their direction of incidence on the sphere. Beamforming methods are therefore applied for their DOA (direction of arrival) estimation. However, the crude applications of these methods revealed significant deviations between the measured and expected DOA. In-depth experiments in anechoic conditions have been carried out to analyze this problem and they revealed that the origin of these deviations could be related to the non-rigidity of the spherical surface of the array. In these experiments, incident waves were generated by a pseudo—point source at several frequencies in the anechoic chamber. The microphone array is rotated and oriented at several azimuthal positions. The sound pressures measured on the sphere by the 16 microphones are then compared with their theoretical values obtained with a rigid sphere assumption, which revealed some differences. Some methods are then proposed to correct this problem. These methods are finally presented and their results in DOA estimation tasks are compared.

Design framework for spherical microphone and loudspeaker arrays in a multiple-input multiple-output system.

Spherical microphone arrays (SMAs) and spherical loudspeaker arrays (SLAs) facilitate the study of room acoustics due to the three-dimensional analysis they provide. More recently, systems that combine both arrays, referred to as multiple-input multiple-output (MIMO) systems, have been proposed due to the added spatial diversity they facilitate. The literature provides frameworks for designing SMAs and SLAs separately, including error analysis from which the operating frequency range (OFR) of an array is defined. However, such a framework does not exist for the joint design of a SMA and a SLA that comprise a MIMO system. This paper develops a design framework for MIMO systems based on a model that addresses errors and highlights the importance of a matched design. Expanding on a free-field assumption, errors are incorporated separately for each array and error bounds are defined, facilitating error analysis for the system. The dependency of the error bounds on the SLA and SMA parameters is studied and it is recommended that parameters should be chosen to assure matched OFRs of the arrays in MIMO system design. A design example is provided, demonstrating the superiority of a matched system over an unmatched system in the synthesis of directional room impulse responses.

A new approach for acoustic visualization using directional impulse response in room acoustics

A new approach is proposed for investigating the temporal and spatial structures of room impulse responses (IRs) using sound field decomposition. A spherical microphone array was used to analyze the directional characteristics of sound fields through reconstructing the sound fields in an actual chamber hall. To perform directional and spatial characterization of recorded sounds, a plane wave decomposition with an adaptive beamforming technique, such as minimum variance distortionless response (MVDR) spatial spectrum estimation, was employed. Based on the proposed method, a directional impulse response (DIR) was generated in the hall, and the sound level was visualized in a video, which was combined with the viewpoint of each receiver. In addition, the DIR was used to calculate the early energy distribution in three-dimensional directions and the results were represented in the hall plan. This method of acoustic visualization can be used to estimate the effects of architectural design components on the directionality of sounds in rooms.

Beamforming with a volumetric array of massless laser spark sources--Application in reflection tracking.

A volumetric array of laser-induced air breakdown sparks is used to produce a directional and steerable acoustic source. The laser breakdown array element is broadband, point-like, and massless. It produces an impulse-like waveform in midair, thus generating accurate spatio-temporal information for acoustic beamforming. A laser-spark scanning setup and the concept of a massless steerable source are presented and evaluated with a cubic array by using an off-line far field delay-and-sum beamforming method. This virtual acoustic array with minimal source influence can, for instance, produce narrow transmission beams to obtain localized and directional impulse response information by reflection tracking.

A new technique to measure directional impulse responses using a 64-channel spherical microphone

A spherical microphone is an array consisting of numerous microphones arranged almost uniformly on the surface of a rigid sphere. Recently, spherical microphone arrays have received attention in architectural acoustics for their ability to characterize the sound field over a full sphere, which can yield additional information about the spatial characteristics of the field, including field diffusivity and directionality of individual reflections. Typically, these arrays are used as real-time analyzers to capture sound pressure information. However, in architectural acoustics, an impulse response measurement is required to adequately characterize the space. An effective approach for utilizing spherical microphones, particularly those with a large number of channels, to capture directional impulse responses has yet to be realized. The current project has utilized a 64-channel spherical microphone to carry out the integrated impulse response measurement technique using a sine-sweep signal. MATLAB has been used to evaluate sound fields in a reverberant chamber using a variety of directionality patterns (e.g., omnidirectional, cardioid, and figure-of-eight). Additionally, work is in progress to allow for arbitrary beamforming and other directionality patterns. Results and analysis will be presented.

Scalable design of microprogrammed digital FIR filter for sensor processing subsystem

In this letter, a novel scalable and modular design of direct form sequential finite impulse response (FIR) filter using microprogrammed control unit is proposed that can be efficiently realized in field programmable gate array (FPGA) or application specific integrated circuit (ASIC). The proposed design is suitable for sensor processing subsystem used in wireless sensor network (WSN) nodes. This is demonstrated by evaluating a sample 4-tap FIR filter on various FPGA platforms and ASIC technologies. The evaluation result shows good area/power efficiency and flexibility by using microprogrammed architecture for such applications.