Room Impulse Responses Research Articles

One acoustician’s defect is another artist’s feature: Simulating real flutter for an art installation

Flutter is generally considered an acoustical defect – the timbral distortions and rhythmic nature can be distracting, annoying, or downright disruptive. The paper recontextualizes flutter as a compositional tool for sound art, particularly when the flutter of existing rooms is used for site-specific installations. Due to the global COVID-19 pandemic, the artist was unable to travel, allowing us to visit the site and take several room impulse responses. The measured spaces were parallelpiped, sound reflective, and fluttery. To give the artist creative flexibility, we simulated the rooms for an extended range of source and receiver locations informed by the in-situ measurements. This paper will discuss our calibration and modeling techniques to simulate flutter and reverberation coloration in real rooms, which is non-trivial for image source methods or ray-based software.

Pneumonia diagnosis algorithm based on room impulse responses using cough sounds

For data augmentation of the pneumonia diagnosis algorithm by deep learning, an image conversion method through a convolutional method utilizing spatial impulse and sound quality factors of cough sounds is proposed. First, reverberant spaces with different volumes were implemented, and spatial impulse responses were generated for each space through computer simulation of spatial models according to sound source and receiver points. Sound quality analysis was performed to improve accuracy, and 2-D sound quality data of time series was converted into 3-D image data using the Gramian Angular Field (GAF) method for combination between heterogeneous data. As a result, 97.5% accuracy was obtained for the configured dataset. The result of this study is expected to be used to develop diagnostic algorithms for various respiratory diseases including pneumonia in the future.

Influence of sound source directivity on finite element simulation of small-room acoustics

Numerical simulation is a flexible and effective method for room acoustic design. Full-range simulation of room acoustics requires a combination of different numerical methods, in which wave acoustic methods (WAM) and geometric acoustic methods (GAM) are used for the low and high frequency region, respectively. In the general low-frequency WAM simulation, a sound source is often assumed to be a point source or loudspeakers are usually approximated by circular planar pistons. However, compared to a large room, the critical frequency between WAM and GAM should be higher in acoustic simulation of small rooms because of relative size between the wavelength and the room, and thus, the above simplifications of actual sound sources, typically loudspeakers, may lead to errors in terms of directivity in the frequency range that the WAM is applied for. Further errors in the desired room impulse responses or other room acoustic parameters caused by directivity errors needs quantitative analysis, and then evaluation can be made on whether it is necessary to consider a more accurate sound source directivity in the WAM simulation of small room acoustics.

RAMP summer bridge: Room acoustics: Connecting research, academics and mentoring

The Research, Academics and Mentoring Pathways (RAMP) to Success summer bridge program at the University of Massachusetts Lowell prepares new engineering students from all majors for an early transition from high school to college. Over the last five years, RAMP has engaged 107 students, the majority being women and underrepresented minorities, in a six-week research, academics and mentoring experience. During RAMP, students are enrolled in Calculus and Introduction to Engineering courses, earning six credits towards their degree pathway. Groups of 5–6 students take on research projects with engineering faculty while also meeting weekly with professionals from industry. The structuring of a room acoustics research project is presented. A project-based learning model helps build skills in collaborative problem solving while also supporting each student to find authentic interest in the research. The project integrates experiments on measuring room impulse responses and spatial variation of source pressure amplitude in rooms of different sizes with a model-driven analysis of the data collected. Concepts of reverberation time, backward integration and room constants reinforce calculus concepts being studied. Coding experiments on rendering speech and music through measured impulse responses and filters created interest across a range of acoustics topics from concert halls to virtual audio.

Perceptual detection thresholds for numerical dispersion in binaural auralizations of two acoustically different rooms.

Room acoustic simulations using the finite-difference time-domain method on a wide frequency range can be computationally expensive and typically contain numerical dispersion. Numerical dispersion can be audible and, thus, constitutes an artifact in auralizations. There is a need to measure perceptual thresholds for numerical dispersion to achieve an optimal balance between computational complexity and audibility of dispersion. This work measures the perceptual detection thresholds for numerical dispersion in binaural auralizations of two acoustically different rooms. Numerical dispersion is incorporated into measured binaural room impulse responses (BRIRs) by the means of filters that represent the dispersion that plane waves experience, which propagate in the simulation in the direction of the worst-case dispersion error. The results show that the perceptual detection threshold is generally lower for the most reverberant room and greatly depends on the source signal independently of the room in which the threshold is measured. It is the most noticeable in the pure BRIRs, i.e., with an impulse as source signal, and almost unnoticeable with speech. The results also show that there was no statistical evidence that the perceptual thresholds for the conditions where numerical dispersion was present or absent in the direct path of the BRIRs be different.

Effect of electric-acoustic cochlear implant stimulation and coding strategies on spatial cues of speech signals in reverberant room

The comparison of spatial cues changes in different setups and coding strategies used in cochlear implants (CI) is investigated. In this experiment, we implement three voice coder setups, such as bilateral CI, bimodal CI, and electro-acoustic stimulation (EAS). Two well-known coding strategies are used, which are continuous interleaved sampling (CIS) and spectral peak (SPEAK). Speech signals are convoluted with appropriate binaural room impulse response (BRIR), creating reverberant spatial stimuli. Five different reverberant conditions (including anechoic) were applied to the stimuli. Interaural level and time differences (ILD and ITD) are evaluated objectively and subjectively, and their relationship with the intelligibility of speech is observed. Prior objective evaluation with CIS reveals that clarity (C50) becomes a more important factor in spatial cue change than reverberation time. Vocoded conditions (bilateral CI) show an increment in ILD value (compression has not been implemented yet on the vocoder processing), when the value of ITD gets more different (decreased) from the middle point. Reverberation degrades the intelligibility rate at various rates depending on the C50 value, both in unvocoded and vocoded conditions. In the vocoded condition, decrement on spatial cues was also followed by the decreement on the intelligibility of spatial stimuli.

Effect of spatial separation of sources on speech intelligibility in an automotive environment

Speech intelligibility has been shown to be enhanced when the target speech source is spatially separated from the interferer, which is related to the spatial unmasking phenomenon. However, this issue has not been studied in the small acoustical space of an automobile. This study, thus, examines the effect of spatial separation of sources on the speech reception threshold (SRT) in an automobile, and compared with the results in a weak-reflective listening room. The target was always presented at the front-passenger seat, and the interferer was presented at the front-passenger seat, right-back seat, mid-back seat, and left-back seat in sequence. The stimuli were synthesized using convolution with binaural room impulse responses measured on a dummy head in driver seat under different interferer locations. Sentence SRTs in Mandarin Chinese were measured via headphones virtually in an automobile and a listening room. Accordingly, the spatial release from masking (SRM) was obtained based on the SRT result. Results show that the SRM in automobile is always smaller than that in listening room, because the early reflections and seat-back occlusions cannot only obscure the binaural cues related to the source localization such as the interaural-time-difference and interaural-level-difference, but also influence the target-to-interferer ratio.

Measuring the Acoustical Properties of the BBC Maida Vale Recording Studios for Virtual Reality

In this paper we present a complete acoustic survey of the British Broadcasting Corporation Maida Vale recording studios. The paper outlines a fast room acoustic measurement framework for capture of spatial impulse response measurements for use in three or six degrees of freedom Virtual Reality rendering. Binaural recordings from a KEMAR dummy head as well as higher order Ambisonic spatial room impulse response measurements taken using a higher order Ambisonic microphone are presented. An acoustic comparison of the studios is discussed, highlighting remarkable similarities across three of the recording spaces despite significant differences in geometry. Finally, a database of the measurements, housing the raw impulse response captures as well as processed spatial room impulse responses is presented.

Convex-optimization-based post-processing for computing room impulse response by frequency-domain FEM

Numerical simulation of impulse responses is an important task for room acoustic design and prediction. Since an impulse response is a concept in the time domain, time-domain simulation methods are popular for this task. At the same time, frequency-domain simulation methods are also used for simulating a room impulse response by computing frequency responses and applying the inverse discrete Fourier transform (IDFT). In this case, well-established frequency-domain solvers, including finite element method (FEM) and boundary element method (BEM), can be used. However, IDFT generates some artifacts that are undesirable for room acoustic evaluation. If such artifacts can be attenuated, then the frequency-domain methods become more attractive for simulating room impulse responses. In this paper, we propose a post-processing method to attenuate the artifacts from the impulse responses computed by the combination of frequency-domain simulation and IDFT. It is formulated as a convex optimization problem, and the alternating direction method of multipliers (ADMM) is applied to obtain the solution. Numerical experiments confirmed that the proposed method can effectively reduce the undesirable artifacts from the reconstructed impulse responses. An example implementation of the proposed method in MATLAB is provided at https://doi.org/g9h5.

Low-latency real-time independent vector analysis using convolutive transfer function

Most blind source separation (BSS) methods use a long short-time Fourier transform (STFT) frame size in highly reverberant environments, which leads to a high algorithmic latency. In addition, the separated signal using the back-projection technology in most BSS methods are source images, and an extra dereverberation processing is required to obtain dereverberated sources. In this paper, we propose a low-latency real-time independent vector analysis (IVA) method using the convolutive transfer function (CTF), where the room impulse response (RIR) is approximately represented by the CTF approximation. The CTF approximation enables representations of long RIRs using a short STFT frame size, and hence the latency of the proposed method is reduced by shortening the STFT frame size in highly reverberant environments. Another advantage of the proposed method is that the dereverberated source can be estimated directly using the derived generalized back-projection technology without extra dereverberation processing. Moreover, the proposed method reduces to the original IVA in the determined case. The proposed method is compared to the state-of-the-art weighted prediction error based IVA, and the results confirm the effectiveness of the presented method in three different reverberation times.

Acoustic modal analysis of room responses from the perspective of state-space balanced realization with application to field interpolation.

Despite its importance in structural dynamics and vibration, modal analysis is rarely performed in acoustics due to the high modal density of sound fields. A novel acoustic modal analysis (AMA) approach is proposed in this paper for enclosed acoustic fields, such as a room, from the perspective of state-space formulation of control systems. A single-input-multiple-output (SIMO) state-space model is established in light of the balanced realization (BR), given impulse response measurements. The BR model is then converted to a modal form such that the modal parameters, including natural frequencies, damping ratios, and mode shapes, can be estimated. To reconstruct mode shapes, plane wave decomposition (PWD) and compressive sensing (CS) techniques are exploited to solve the underdetermined problem for a spatially sparse representation of mode shapes under the Schroeder frequency. As a result, a model of a continuous system can be "interpolated" for any arbitrary source-receiver positions on the basis of the estimated mode shapes. With the identified modal parameters, the low-frequency and early reflection part of room impulse responses (RIRs) can be synthesized for arbitrary source-sensor pairs. The proposed AMA acoustic field interpolation is validated by extensive simulations and experiments.

The Optimal Determination of the Truncation Time of Non-Exponential Sound Decays

The noise effects in the room impulse response (RIR) make the decay range of the integrated impulse response insufficient for reliable determination of reverberation time (RT). One of the preferred techniques to minimize noise effects is based on noise subtraction, RIR truncation, and correction for the truncation. The success of RT estimation through the method depends critically on the accurate estimation of the truncation time (TT). However, noise fluctuation and RIR irregularities can lead to discrepancies in the determined TT from the optimal value. The general goal of this paper is to improve RT estimates. An iterative procedure based on a non-exponential decay model consisting of a double-slope decay term and a noise term is presented to estimate the TT accurately. The model parameters are generated until the iterative procedure converges to a minimum difference between the energy decay curve (EDC) generated by the model and the Schroeder decay function. The decay rates of the EDCs with added pink noise levels are compared to those of the EDCs with low background noise. In addition, the detected TTs and the corresponding RTs are compared with the existing method and the noise compensation method (subtraction–truncation–correction method).

Time–frequency methods for characterization of room impulse responses and decay time measurement

In the low frequency range, the reverberation of a room should be characterized by the decay time, which can be determined experimentally by various methods. In order to make accurate and precise measurements, the differences between these methods and their advantages should be known and quantified. In this work, the performance of four linear and four adaptive time–frequency impulse response analysis methods was evaluated with the aim to find the most accurate method for characterizing the decays of the room modes. The methods were compared using generated Prony test signals with known modal properties and measured room impulse responses were also examined. The decay time using the selected time–frequency impulse response method was then estimated in three rooms and the measurement results were compared with interrupted sine tone methods. It has been shown that the choice of measurement setup and analysis parameters has an important influence on the estimation of the decay time and its modal frequency. The time–frequency impulse response method using the window width optimized Stockwell transform, the continuous wavelet transform, or the Morlet wave method were found to be the most robust methods, and their applicability is also enhanced by the fact that only a single measurement is required for the selected microphone position. The presented test methods could be used to evaluate the performance of other approaches for the evaluation of reverberation below the Schroeder frequency or the ability of measurement and signal processing techniques to characterize low frequency transient signals.

Effect of listener head orientation on speech reception threshold in an automotive environment

The head orientation of the listener significantly affects speech intelligibility (SI) in automobiles due to the effect of binaural listening and special acoustic conditions such as early reflections and seat-back occlusions. However, this issue has not been studied with subjective tests yet. This study investigates SI with various head orientations of a listener in automobiles with an subjective experiment. The sentence speech reception thresholds (SRT) in Mandarin Chinese are measured via headphones virtually in an automobile environment, and compared with results in a weak-reflective listening room. A virtual speaker is located in the front-passenger seat, the right-back seat and the left-back seat in sequence, by convolving the target speech with corresponding binaural room impulse responses measured on a dummy head in the driver’s seat with five head orientations. Result shows that the SRT variations caused by head orientations are up to 5 dB in automobile, lower than that of 9 dB in listening room. Under various head orientations, the SRT in automobile decreases as the virtual speaker moves closer to the front lateral direction, which is similar to the change rule in listening room that is mainly determined by the effect of binaural listening. Overall, a lower SRT in automobile can be obtained when the listener in the driver seat turns their head inward, i.e., towards the right. In comparison with the result in listening room, early reflections improve SI in automobile by an SRT decrease of up to 3.5 dB, while seat-back occlusions reduce SI by an SRT increase of up to 5.3 dB. The early reflections play a more significant role when the listener is in an adverse position, e.g., for the head orientations and speaker locations making direct sounds difficult to reach the listener’s ears. Comparison between the SRTs based on speech transmission index (STI) and STI-SI models and the SRTs measured by subjective experiment indicates that the STI-based objective method only partially expresses the variation in SI obtained from subjective experiment, and the difference between the subjective results and the objective results is mainly caused by the inapplicability of the STI-SI models derived from the traditional room in the automotive environment. The present work is relevant to understanding the combined effect of various factors on SI under such a special acoustic condition in automobile.

Deep Prior Approach for Room Impulse Response Reconstruction

In this paper, we propose a data-driven approach for the reconstruction of unknown room impulse responses (RIRs) based on the deep prior paradigm. We formulate RIR reconstruction as an inverse problem. More specifically, a convolutional neural network (CNN) is employed prior, in order to obtain a regularized solution to the RIR reconstruction problem for uniform linear arrays. This approach allows us to avoid assumptions on sound wave propagation, acoustic environment, or measuring setting made in state-of-the-art RIR reconstruction algorithms. Moreover, differently from classical deep learning solutions in the literature, the deep prior approach employs a per-element training. Therefore, the proposed method does not require training data sets, and it can be applied to RIRs independently from available data or environments. Results on simulated data demonstrate that the proposed technique is able to provide accurate results in a wide range of scenarios, including variable direction of arrival of the source, room , and SNR at the sensors. The devised technique is also applied to real measurements, resulting in accurate RIR reconstruction and robustness to noise compared to state-of-the-art solutions.

Numerical modeling and verification of room impulse response inference from the coherence properties of speech and music signals

In previous work, we presented theory and motivating examples to demonstrate that the coherence properties of the harmonic partials of speech and music signals recorded in an acoustic space give information about the impulse response of the space. By analyzing auto- and cross-correlations between pairs of harmonics of the signal, we may “blindly” obtain the frequency-dependent reverberation time of the acoustic space. Assuming that the signal contains harmonically related partials with mutually correlated amplitude and phase modulations, a reasonable assumption for sources such as musical instruments or the human voice, this approach allows inference of information about the impulse response of an acoustic space. In this work, we verify this method for a variety of signals with different coherence times and amounts of pseudo-random amplitude and frequency modulations.

Spatial room impulse response measurement and post-processing for virtual reproduction of a stereo sound system

Spatial room impulse responses (SRIRs) characterize the spatial, temporal, and spectral properties of the sound field received at a particular location in a room for a given source-receiver configuration. SRIRs can be captured using spherical microphone arrays and are useful both for evaluating the acoustic performance of a space and in applications where it is desired to auralize the sound field being characterized with varying signals being played from the source. In this work, SRIRs were measured in a living room (mid-frequency RT approximately 0.4 s) using an mh acoustics Eigenmike em32 as the receiver and both a studio monitor loudspeaker and a subwoofer as the sources. Studio monitor placements replicated a left-right-center stereo monitoring configuration, with the subwoofer on the floor below the center loudspeaker, while attempting to avoid source and receiver placement on estimated modal planes in the room. Details of the post-processing used to finalize the SRIRs for use as a virtual stereo sound system will be presented, including cross-over filtering, tuning, and correction of the overall frequency response to a curve that is favorable for in-room loudspeaker listening. Measurement system details, SRIR data visualizations, and RT estimates will also be discussed.

Impact Thresholds of Parameters of Binaural Room Impulse Responses (BRIRs) on Perceptual Reverberation

This paper presents a study on the perceived importance of different acoustic parameters of Binaural Room Impulse Response (BRIR) rendering. A headphone-based listening test was conducted with twenty expert participants. Three BRIRs generated from simulations of three different rooms were convolved with a dry speech signal and used as reference audio samples. Four BRIR parameters, Initial Time Delay Gap (ITDG), Forward Early Reflections (FER), Reverse Early Reflections (RER) and Late Reverberation (LR) were systematically altered and convolved with a speech signal to generate the test conditions. A staircase method was used to obtain the threshold at which each BRIR parameter was perceived as different from the reference audio sample. The average perceived impact threshold of each parameter was then calculated across the twenty participants. Results show that RER removal and ITDG extension have a clear impact on the perceptual reverberation of speech audio. Subjects were less sensitive to FER removal. The effect of LR removal on perceptual reverberation is hard to distinguish. Therefore, RER and ITDG are of particular importance when designing artificial reverberation algorithms, whilst more research is needed to understand the perceptual contribution of LR. Minor changes in FER and LR are less significant.

Favorable reverberation time in concert halls revisited for piano and violin solos.

A favorable reverberation time in concert halls is a fundamental issue for room acoustic design, and various recommendations have been suggested so far. Nevertheless, one must track back more than half a century when it comes to systematic subjective experiments on this topic. In this study, binaural room impulse responses (RIRs) measured with a dodecahedral loudspeaker in concert halls, where orchestra concerts are regularly held, are used. First, signal processing is applied to RIR to equalize the dodecahedral loudspeaker response as flat and broad as possible within the audible frequency range. Then, anechoic recordings of music by piano and violin, excerpts from cadenzas, are convoluted with the equalized RIR. Next, subjective experiments are conducted to seek favorable reverberation times. Reverberance and clarity were judged by 16 music experts. From this research, the favorable reverberation times RTM (octave band average for 500 and 1000 Hz) for piano and violin solos are from 1.2 to 2.0 s and 1.8 to 2.4 s, respectively. However, the clarity index C80,3 (octave band average for 500, 1000, and 2000 Hz) needs to range from 0 to 2.4 dB and -1.6 to 0.7 dB, respectively, to meet the optimum reverberance for piano and violin.

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.