Impulse Response Measurements Research Articles

Sound propagation modelling in a street canyon by combining an image source model and impulse response measurements

This research introduces an innovative approach to achieve a computationally efficient model for calculating impulse responses in an urban environment, more precisely a street canyon, for auralization purposes. The image source method has shown its merits before, but when used for auralization, it typically lacks realism. In this study the image source method is used to create a basic impulse response of a street canyon, enriched by a stochastic response based on an envelope function. While the image source model accounts for the most prominent reflections, the stochastic part is due to all scattered sound energy in the street originating mainly from facade irregularities and street furniture. To obtain realistic frequency dependent envelope functions for various street canyon geometries, the function is based on measured impulse responses in nearly 100 inner-city street canyons in the city of Ghent, which were conducted by Thomas et al. (2013). By the proposed approach, we are able to generate impulse responses for street canyons suitable for sound perception research in an efficient way. More studies are needed to verify the applicability of the model for streets outside the Ghent dataset.

Characterization of façade sound insulation by impulse response measurements for various source position formats: a case study

The sound insulation of a building's façade is important for estimating the indoor levels in relation to the outdoor levels. The measurement of the sound insulation (i.e., outdoor-to-indoor level difference) is significantly affected by the methodology employed. In this study, an impulse response measurement methodology is incorporated, alleviating uncertainties associated with the measurement's chain. The effect of loudspeaker position on sound insulation is investigated with respect to three source position formats (horizontal, inclined, and linear), each consisting of three source positions. The appropriate length of the excitation signal is verified using the impulse-to-noise ratio (INR) indicator. Focusing on the outdoor-to-indoor level differences, corresponding to the energy captured in each external receiver (placed very close to windows) and across all the internal receivers per source position, significant differences were found. This indicates the impact of both the distance and angle of incidence between the outdoor receiver and the sound source position. In terms of the average outdoor-to-indoor level difference among the three source formats, the linear format shows slightly lower values compared to the horizontal and inclination source formats, which are similar to each other, but follows the main trend.

Acoustic behaviour and acoustic treatment strategies for the baroque church of Santuari del Miracle, Solsonès, Lleida

The church of Santuari del Miracle, located in the Catalan region of Solsonès, Lleida, is a unique barroque church in Catalonia that hosts Baroque music performances, like the festival Espurnes Barroques. However, according to the musicians and the audience, it is too reverberant to play this style of music. The objective of this work is to examine the current acoustic performance of the church and to propose acoustic treatment strategies that are respectful of this unique heritage building. Impulse response measurements are first performed following ISO 3382-1 standards, using Odeon software to assess reverberation time (T30), Early Decay Time (EDT), Clarity (C50), and Definition (D50). Simulations are next performed using Odeon. They are used to refine the smallest perceptible differences from the measured data and to examine potential treatment strategies. The proposed solution consists of elements based on variable acoustics that can be easily hidden or removed, thus preserving the original acoustics of this heritage building.

Exploring sauna impulse responses

Sauna is an important element of Finnish tradition and culture, and its properties and features deserve preservation and archiving. Additionally, extreme temperature and humidity in a sauna comprise a unique environment for researching room acoustics in unusual atmospheric conditions. In the present study, we publish a dataset of room impulse response (RIR) measurements of a sauna during the process of warming up and throwing water on the stove. We explore several features of the dataset, such as the impact of the temperature and humidity distributions on the time-of-arrival of reflections, the atmospheric absorption of sound, and spectral coloration of measured RIRs. The results of this research show the influence of environmental factors on RIRs and allow an insight into the distinct room acoustics of a sauna while heating up.

Modal identification as a tool to increase the accuracy of acoustic absorption measurements in reverberation rooms

Measuring the acoustic properties of materials may be very challenging. At low frequencies, the sample may interfere with the room's modal behavior. When it occurs, the longest modal decays are influenced. Differences in reverberation time - empty Vs with-specimen conditions - are often emphasized by these modal effects, and, as a matter of fact, the absorption coefficient may be overestimated. The present work aims to increase the accuracy through data augmentation at low frequencies. Modal identification was performed on measured impulse responses of porous specimens, allowing the determination of modal-decay times within the third-octave bands of 80-160 Hz. A statistical analysis of natural modes highlights the modes whose decay exceeds the density function as outliers. This preliminary study ultimately provides some statistical considerations on the modal distribution of decay times and their relationship with the reverberation time measured through the Schroeder integral.

Virtual sound source construction based on adaptive crossfade processing with electro-dynamic and parametric loudspeaker arrays

Among the various high-presence sound field reproduction technologies, we focus on wave field synthesis (WFS) using a loudspeaker array, as it can construct a virtual sound source (VSS) with few restrictions. A VSS using parametric array loudspeakers (PALs) can provide more accurate direction perception than VSS using electro-dynamic loudspeakers (EDLs) because PALs hold sharp directivity by utilizing ultrasounds. However, the distance perception tends to be degraded. We previously proposed a VSS construction based on WFS with hybrid control of an EDL and a PAL arrays. This method enhances distance perception by exploiting the difference in radiation characteristics between PALs and EDLs. However, the difference in frequency responses between PAL and EDL is noticeable to the listener when switching between them. In this paper, to enable more smooth transition of the EDL and PAL arrays, we propose a VSS construction method based on adaptive crossfade processing with EDL and PAL arrays. For an arbitrary VSS construction position, the coefficients of crossfade for each frequency are determined by an adaptive filter based on measured impulse responses of VSSs with EDL and PAL arrays. We demonstrated the effectiveness of the proposed method through evaluation experiments.

Low-frequency chatter suppression for robotic milling using a novel MRF absorber

Robotic milling has a unique advantage for large and complex parts. However, it is extremely prone to low-frequency chatter due to the robot structure mode. In robotic milling, low-frequency chatter has a huge impact on machining quality and efficiency. In this paper, a novel MRF (Magnetorheological fluid) absorber is used to suppress low-frequency chatter during robotic milling based on the proposed low-frequency chatter suppression strategy. First, the GPR (Gaussian process regression) model is used to predict the impulse response function based on a few measured impulse response functions. Next, a cutting force model is developed considering the tool runout and the robot structure mode. The parameters and coefficients of the cutting force model are optimized by the measured cutting forces. Then, the low-frequency chatter frequency is predicted considering the modal coupling effect based on the predicted impulse response function and the simulated cutting force. Finally, an MRF absorber is designed based on the chatter frequency range. The controlled storage modulus of the MRF in the pre-yield region enables a wider frequency shift characteristic of the designed MRF absorber. The MRF absorber controls the input current based on the predicted low-frequency chatter frequency to achieve chatter suppression during robotic milling. Robotic milling experiments verified that the MRF absorber can effectively suppress low-frequency chatter. The experimental results show that the Energy Ratio Chatter Indicator of the industrial robot with MRF absorber is less than 0.1 under different milling conditions.

Ionospheric Channel Impulse Response Measurement System for NVIS Propagation Mode Over Java Island Based on Low‐Cost SDR Platform

AbstractThe development of a digital high‐frequency (HF) radio communication system requires an ionospheric channel model from the channel impulse response (CIR) measurement. Although the Watterson ionosphere channel model has been available and used for a long time, several CIR measurements have been conducted in all regions of the Earth in an attempt to validate or replace the Watterson channel model with a suitable model for their region. However, only a few CIR measurements were conducted in low‐latitude regions, especially over Indonesia. In this study, we develop the CIR measurement system for the near vertical incidence skywave (NVIS) propagation mode over Java Island based on the software defined radio platform to meet low‐cost and simple operation requirements. The specification of the system is based on the International Telecommunication Union ionospheric channel characteristic document and increased in order to be able to capture higher values. Results from a 1‐week campaign measurement period show the ability of the system to measure the root mean square of time delay within the range of 0.2–1.3 ms and the Doppler shift within the range of 0.7–1.1 Hz in the quiet conditions of the ionosphere. Further measurements will be conducted to obtain a comprehensive ionosphere CIR that is useful for designing the NVIS‐HF digital communication in Indonesia, which is located beneath the crest region of an equatorial ionospheric anomaly.

A novel approach for extracting contact impedance of electrode/electrolyte interface by minimizing error between impulse response and system function

Electrochemical impedance spectroscopy (EIS) is widely used to analyze the electrode/electrolyte interface, but it has the disadvantage of long measurement time. Previous studies reduced this latency by utilizing specific waveforms such as square and dynamic signals. However, these methods can easily lose phase information due to the measurement noise and constrained sampling rate, which compromises the accuracy of impedance calculations. Here, we propose a novel approach to rapidly extract electrochemical properties at the electrode/electrolyte interface using only the magnitude information of measured signals. The proposed method obtains the contact impedance by minimizing the magnitude error between the theoretical system function and the measured impulse response, demonstrating greater resilience to the phase loss. When calculating the difference rate with conventional EIS, the extracted contact resistance and capacitance data exhibit a 10.34 % and 59.24 % improvement, respectively, compared to the previous approach that utilizes the Nyquist plot incorporating phase data.

A Signal-Processing-Based Simulation System for High-End Stereo Headsets.

In recent years, headphones have become increasingly popular worldwide. There are numerous models on the market today, varying in technical characteristics and offering different listening experiences. This article presents an application for simulating the sound response of specific headphone models by physically wearing others. In the future, for example, this application could help to guide people who already own a pair of headphones during the decision-making process of purchasing a new headphone model. However, the potential fields of application are much broader. An in-depth study of digital signal processing was carried out with the implementation of a computational model. Prior to this, an analysis was performed on impulse response measurements of specific headphones, which allowed for a better understanding of the behavior of each set of headphones. Finally, an evaluation of the entire system was conducted through a listening test. The analysis of the results showed that the software works reasonably well in replicating the target headphones. We hope that this work will stimulate further efforts in the same direction.

Enhancement of virtual acoustics rendering using boundary mounted dipole loudspeakers

The Immersive Media Lab at McGill University hosts a Virtual Acoustics Technology (VAT) system incorporating a suspended array of omnidirectional loudspeakers. Using a convolution reverb engine, acoustic simulations of real spaces can be realized via a catalog of Room Impulse Response measurements. Reflected sound in the room helps to disguise the location of the sound emitters rendering virtual acoustics. One limitation of the system, however, is the interference between the lab’s natural acoustics and the virtual environment generated by the VAT system. The improvement under consideration is to enhance diffusion along the walls of the lab, in order to mask the acoustical characteristics related to the physical dimensions of the room. Dipole loudspeakers are installed on the room boundaries and used to scatter reflections and reverberation along the wall surfaces enlarging the effective radiation surface of the walls. The scattered energy may mask specular reflections and reduce localization of the virtual acoustic sources. Investigations compare the result of scattering reflected sound vertically as opposed to horizontally across the boundaries. Measurements illustrate the effect of the dipole loudspeaker system when used on its own as well as working in conjunction with the existing omnidirectional loudspeaker array.

Increasing the accuracy of ISO 354 and ASTM C423 through modal identification

Measuring the acoustic properties of materials is a highly challenging task. Previous studies have highlighted errors in the absorption coefficient above 200 Hz, leading to absorption coefficient values higher than one. These errors are often attributed to differences in modal decays with and without the specimen. This effect is amplified in the case of highly reactive materials, such as innovative materials, that significantly impact the acoustic field and, consequently, the modal behaviour of the room. This work aims to reduce the difference between measured reverberation time values, with and without the specimen, through Data Augmentation. Modal identification was performed on measured impulse responses of porous specimens, where the alpha is known, allowing the determination of modal decay times and reverberation times within the frequency range of 70–180 Hz. A statistical analysis of natural modes and numeric solutions enables the identification of outliers. In this way, even in a non-well-optimized reverberation room, it is possible to enhance the accuracy of acoustic absorption measurements.

Vibrational Analysis of a Splash Cymbal by Experimental Measurements and Parametric CAD-FEM Simulations

The present study encompasses a thorough analysis of the vibrations in a splash musical cymbal. The analysis is performed using a hybrid methodology that combines experimental measurements with parametric computer-aided design and finite element method simulations. Experimental measurements, including electronic speckle pattern interferometry, and impulse response measurements are conducted. The interferometric measurements are used as a reference for the evaluation of finite element method modal analysis results. The modal damping ratio is calculated via the impulse response measurements and is adopted by the corresponding simulations. Two different approximations are employed for the computer-aided design and finite element method models: one using three-point arcs and the other using lines to describe the non-smooth curvature introduced during manufacturing finishing procedures. The numerical models employing the latter approximation exhibit better agreement with experimental results. The numerical results demonstrate that the cymbal geometrical characteristics, such as the non-smooth curvature and thickness, greatly affect the vibrational behavior of the percussion instrument. These results are of valuable importance for the development of vibroacoustic numerical models that will accurately simulate the sound synthesis of cymbals.

Performance of low-frequency sound zones with very fast room impulse response measurements.

Sound zone methods aim to control the sound field produced by an array of loudspeakers to render a given audio content in specific areas while making it almost inaudible in others. At low frequencies, control filters are based on information of the electro-acoustical path between loudspeakers and listening areas, contained in the room impulse responses (RIRs). This information can be acquired wirelessly through ubiquitous networks of microphones. In that case and for real-time applications in general, short acquisition and processing times are critical. In addition, limiting the amount of data that should be retrieved and processed can also reduce computational demands. Furthermore, such a framework would enable fast adaptation of control filters in changing acoustic environments. This work explores reducing the amount of time and information required to compute control filters when rendering and updating low-frequency sound zones. Using real RIR measurements, it is demonstrated that in some standard acoustic rooms, acquisition times on the order of a few hundred milliseconds are sufficient for accurately rendering sound zones. Moreover, an additional amount of information can be removed from the acquired RIRs without degrading the performance.

Drone audition: improved Gaussian mixture model Wiener filtering approach for audio signal enhancement

Audio signal enhancement is an integral function of drone audition systems which emerges intelligent audio applications for military and civil environments. Signal enhancement using a drone is challenging due to its emission of significant noise from drone motors and propellers resulting in an extremely low signal-to-drone noise ratio level. A method for signal enhancement from an on-board microphone array on a drone for static flight conditions in adverse noisy environments is proposed. In this paper, we present an improved Gaussian mixture model Wiener filter approach for a noisy drone platform in the presence of interfering noise sources such as wind, and traffic. The key to this method is to exploit the spatial properties of the noise sources by using a set of Gaussian components to derive an accurate estimate of the power spectral densities of both residual drone noise and interfering noise sources to obtain the enhanced target sound source signal. We demonstrate the method's general applicability for both speech, and bird calls using an extensive experimental study, including an experimental recording from a drone acoustics dataset using measured impulse responses, and an outdoor measurement from a hovering drone for a bioacoustic application, respectively.

Direct sound field estimation based on sound source modeling with sparse equivalent sources

The measurement of multi-point room impulse responses (RIRs) is challenging because it requires many microphones or a microphone traverse system. In our previous study, we proposed a spatial extrapolation method to measure early RIRs by the locally-positioned microphones. However, the estimation accuracy was degraded at a far distance from the microphones owing to the directivity of the sound source. In this study, we proposed a modeling method to estimate a direct sound field using a source radiation model based on the sparse equivalent source method. First, the source radiation was modeled using an enclosed microphone array. Then, we modeled the direct sound field using locally positioned microphones. The evaluation experiment was conducted in an anechoic chamber under 2.5D condition.

Room acoustic computer modeling—A case study to compare industry standard software with measured results

This paper extends a previous publication presented at the Inter-Noise 2005 conference [1] and investigates the predictive performance of some industry standard room acoustic software packages compared with measurements. The case study specifically examines the Twyford Theatre, comparing predictions of room acoustic parameters obtained from ODEON, EASE, and Treble software with measurements. To conduct the study, impulse response measurements were performed in the theater using the swept sine method. Architectural drawings and on-site inspection measurements were utilized to construct ODEON, EASE, and Treble models for prediction purposes. The findings of this study provide insights into the predictive performance of these industry standard room acoustic software packages with comparison to measured results. C. Field and S. Shimada, Acoustic Computer Modelling: A Case Study to Compare Predictions by CATT and Odeon with Measured Results, Inter-Noise 2005.

A binaural room impulse response dataset and Shorelining psychophysical task for the evaluation of auditory sensory augmentation

Sensory augmentation using spatial sound presented in augmented-reality (AR) can assist people with low vision or blindness in navigating their environment (Katz et al., 2012). Nonetheless, in many situations, the poor quality of the binaural sound rendered using current tool sets limits the potential capability of the assistive technology. In particular, acoustic environments with near-field sources and reflections pose significant challenges. In this work, we provide a reference binaural room impulse response (BRIR) dataset with near-field sources and reflections and an associated shorelining psychophysical task that is useful for the evaluation of AR spatial audio. The dataset consists of 12∼small loudspeakers arranged on a 3-by-4 grid in a complex acoustic environment. BRIR measurements are recorded using the Head and Torso Simulator (HATS) for 17∼different receiver positions with a 5o angular resolution. Room impulse response measurements are also recorded using the Eigenmike for each of the 17 receiver positions. Using the Razer Anzu smart glasses to render the binaural AR spatial audio, we compare psychophysical performance on the shorelining navigation task using the recorded dataset and various existing binaural AR tool sets.

Identification of Solid and Liquid Materials Using Acoustic Signals and Frequency-Graph Features.

Material identification is playing an increasingly important role in various sectors such as industry, petrochemical, mining, and in our daily lives. In recent years, material identification has been utilized for security checks, waste sorting, etc. However, current methods for identifying materials require direct contact with the target and specialized equipment that can be costly, bulky, and not easily portable. Past proposals for addressing this limitation relied on non-contact material identification methods, such as Wi-Fi-based and radar-based material identification methods, which can identify materials with high accuracy without physical contact; however, they are not easily integrated into portable devices. This paper introduces a novel non-contact material identification based on acoustic signals. Different from previous work, our design leverages the built-in microphone and speaker of smartphones as the transceiver to identify target materials. The fundamental idea of our design is that acoustic signals, when propagated through different materials, reach the receiver via multiple paths, producing distinct multipath profiles. These profiles can serve as fingerprints for material identification. We captured and extracted them using acoustic signals, calculated channel impulse response (CIR) measurements, and then extracted image features from the time-frequency domain feature graphs, including histogram of oriented gradient (HOG) and gray-level co-occurrence matrix (GLCM) image features. Furthermore, we adopted the error-correcting output code (ECOC) learning method combined with the majority voting method to identify target materials. We built a prototype for this paper using three mobile phones based on the Android platform. The results from three different solid and liquid materials in varied multipath environments reveal that our design can achieve average identification accuracies of 90% and 97%.

Effects of landscape and distance in automatic audio based bird species identification.

The present work focuses on how the landscape and distance between a bird and an audio recording unit affect automatic species identification. Moreover, it is shown that automatic species identification can be improved by taking into account the effects of landscape and distance. The proposed method uses measurements of impulse responses between the sound source and the recorder. These impulse responses, characterizing the effect of a landscape, can be measured in the real environment, after which they can be convolved with any number of recorded bird sounds to modify an existing set of bird sound recordings. The method is demonstrated using autonomous recording units on an open field and in two different types of forests, varying the distance between the sound source and the recorder. Species identification accuracy improves significantly when the landscape and distance effect is taken into account when building the classification model. The method is demonstrated using bird sounds, but the approach is applicable to other animal and non-animal vocalizations as well.