Room Acoustic Environments Research Articles

How room acoustics impact speech comprehension by listeners with varying English proficiency levels

The authors previously reported the preliminary results from an investigation on effects of reverberation and noise on speech comprehension by native and non-native English-speaking listeners at ICA, Montreal. The results showed significant main effects of reverberation time (from 0.4 to 1.2 s) and background noise level (three settings of RC-30, 40, and 50). Non-native listeners performed significantly worse than natives on speech comprehension in general. Furthermore, the negative effect of reverberation was more detrimental for non-native than for native English-speaking listeners. However, the preliminary analyses have not yet accounted for effects of English proficiency on speech comprehension in addition to the room acoustic environments tested. All test participants were screened for three measures of English proficiency (listening span, oral comprehension, and verbal abilities). Non-native listeners as a group scored lower on all three proficiency measures than native English-speaking listeners. In this paper, English proficiency is investigated as confounding factor affecting speech comprehension performance alongside noise and reverberation. The results help to further the understanding of how room acoustics impacts speech comprehension by listeners, native and non-native English-speaking, with varying English proficiency levels. [Work supported by a UNL Durham School Seed Grant and the Paul S. Veneklasen Research Foundation.]

Time delay estimation via non-mutual information among multiple microphones

Time delay estimation (TDE) is an important part of sound source localization in hands-free communication systems. TDE in room acoustic environments is usually affected by reverberation. To improve the robustness of time delay estimators to reverberation, a novel TDE algorithm is proposed from information theory perspective. The non-mutual information among multiple microphone signals is extracted to estimate the time delay. Superiority of the proposed algorithm is demonstrated in reverberant environments.

Time Difference of Arrival Estimation Exploiting Multichannel Spatio-Temporal Prediction

To localize sound sources in room acoustic environments, time differences of arrival (TDOA) between two or more microphone signals must be determined. This problem is often referred to as time delay estimation (TDE). The multichannel cross-correlation-coefficient (MCCC) algorithm, which is an extension of the traditional cross-correlation method from two- to multiple-channel cases, exploits spatial information among multiple microphones to improve the robustness of TDE. In this paper, we propose a multichannel spatio-temporal prediction (MCSTP) algorithm, which can be viewed as a generalization of the MCCC principle from using only spatial information to using both spatial and temporal information. A recursive version of this new algorithm is then developed, which can achieve similar performance as MCSTP, but is computationally more efficient. Experimental results in reverberant and noisy environments demonstrate the advantages of this new method for TDE.

The role of acoustical characteristics for enhancement of acoustic comfort in high-speed train passenger cars

The room acoustic environments in high-speed trains have been investigated to identify the design elements for the passenger cars to improve acoustic comfort. Both room acoustical and psychoacoustical parameters affected by the absorption coefficients of interior finish materials were measured at the height of passengers’ ears. Room acoustical simulation was constructed based on the measurements to investigate the effect of design elements influencing acoustic quality in the carriage. Through computer simulation of the models with changes in acoustical properties such as absorption/diffusion coefficients of the interior surfaces, the effect of interior design components were investigated and classified to improve the speech privacy.

Automatic speech recognition performance in different room acoustic environments with and without dereverberation preprocessing

The performance of recent dereverberation methods for reverberant speech preprocessing prior to Automatic Speech Recognition (ASR) is compared for an extensive range of room and source-receiver configurations. It is shown that room acoustic parameters such as the clarity (C50) and the definition (D50) correlate well with the ASR results. When available, such room acoustic parameters can provide insight into reverberant speech ASR performance and potential improvement via dereverberation preprocessing. It is also shown that the application of a recent dereverberation method based on perceptual modelling can be used in the above context and achieve significant Phone Recognition (PR) improvement, especially under highly reverberant conditions.

Localization of distinct reflections in rooms using spherical microphone array eigenbeam processing

This paper presents an experimental and comparative study of several spherical microphone array eigenbeam (EB) processing techniques for localization of early reflections in room acoustic environments, which is a relevant research topic in both audio signal processing and room acoustics. This paper focuses on steered beamformer-based and subspace-based localization techniques implemented in the spherical EB domain, including the plane-wave decomposition, eigenbeam delay and sum, eigenbeam minimum variance distortionless response, eigenbeam multiple signal classification (EB-MUSIC), and eigenbeam estimation of signal parameters via rotational invariance techniques (EB-ESPRIT) methods. The directions of arrival of the original sound source and the associated reflection signals in acoustic environments are estimated from acoustic maps of the rooms, which are obtained using a spherical microphone array. The EB-domain-based frequency smoothing and white noise gain control techniques are derived and employed to improve the performance and robustness of reflection localization. The applicability of the presented methods in practice is confirmed by experiments carried out in real rooms.

A system for simulating room acoustical environments for one’s own voice

The real-time simulation of room acoustical environments for one’s own voice using generic software has been difficult until very recently due to the computational load involved: requiring real-time convolution of a person’s voice with a potentially large number of long room impulse responses. This paper describes a software-based solution that accomplishes real-time convolution with head-tracking to simulate the effect of room acoustical environments on the sound of one’s own voice, using binaural technology. Actual rooms are characterized by measuring the room impulse response from the mouth to ears of the same head (oral binaural room impulse response, OBRIR). By repeating this process at 2° yaw increments for a given head position, the rooms are binaurally scanned around a given position to obtain a collection of OBRIRs, which is then used by the software-based simulation system. In the simulated rooms, a person equipped with a near-mouth microphone and near-ear loudspeakers can speak or sing and hear their voice, as it would sound in the recorded rooms, while physically being in an anechoic room. By continually updating the person’s head orientation using head-tracking, the corresponding OBRIR is chosen for convolution with their voice. The system described in this paper achieves the low latency that is required to simulate nearby reflections, and it can perform convolution with long room impulse responses.

Enhanced speech separation in room acoustic environments with selected binaural cues

We propose a robust technique to separate audio sources received by a microphone array in a room acoustic environment with an underdetermined mixing process (i.e., the number of sources is larger than the number of mixtures). Our scheme consists of two stages: 1) estimation of mixing parameters and 2) recovery of source signals. For the first stage, contrary to the traditional DUET-type methods that exploit all binaural cues, we estimate the mixing parameters by selecting a reliable subset of binaural cues based on the phase determinacy condition and source sparsity. As a result, we can determine the mixing parameters successfully even in a reverberant environment with longer time delay. Then, proper mathematical tools are applied to the underdetermined linear system to recover the original audio sources for the second stage. Experimental results on simulated data in a room acoustic environment are given to show a significant gain over the DUET-type method in audio source separation.

An investigation of time delay estimation in room acoustic environment using magnitude ratio

This paper investigates the relation between the nonstationary sound source and the frequency domain magnitude ratio of two microphones based on short-term frequency analysis. The fluctuation level of nonstationary sound sources is modeled by the exponent of polynomials from the concept of moving pole model. According to this model, the sufficient condition for utilizing the fluctuation level and magnitude ratio to estimate the time delay between two microphones is suggested. Simulation results are presented to show the performance of the suggested method.

Multi-rate Updated Adaptive Algorithm Applied to Acoustic Echo Cancellation

By analysing the actual room acoustic environment, this paper presents a novel algorithm for acoustic echo cancel- lation,which can dynamically change the coefficients update- rate of the adaptive filter.This algorithm builds up a nonlinear function relationship between update-rate and the error signal. The update-rate of the coefficients increases automatically when the environment varies,while it would be smaller in a stable state,which can lead to significant reduction in the computa- tional burden.Theoretical analysis and computer simulations indicate that the convergence and tracking properties of this al- gerithm are as good as the traditional algorithms.

Time Delay Estimation in Room Acoustic Environments: An Overview

Time delay estimation has been a research topic of significant practical importance in many fields (radar, sonar, seismology, geophysics, ultrasonics, hands-free communications, etc.). It is a first stage that feeds into subsequent processing blocks for identifying, localizing, and tracking radiating sources. This area has made remarkable advances in the past few decades, and is continuing to progress, with an aim to create processors that are tolerant to both noise and reverberation. This paper presents a systematic overview of the state-of-the-art of time-delay-estimation algorithms ranging from the simple cross-correlation method to the advanced blind channel identification based techniques. We discuss the pros and cons of each individual algorithm, and outline their inherent relationships. We also provide experimental results to illustrate their performance differences in room acoustic environments where reverberation and noise are commonly encountered.

A specialized fast cross-correlation for acoustical measurements using coded sequences

In acoustics applications, binary maximal-length sequences and related sequences are increasingly used for acoustics system identification tasks. A number of coded sequences, such as binary maximal-length related sequences and ternary sequences possess two-valued or pulselike autocorrelation functions. It is this correlation property that is exploited in most of acoustical applications. However, the length of some of these sequences is not directly suitable for FFT-based cross-correlation algorithms. This paper explores using standard FFTs to calculate the cross-correlation between two periodic finite-length sequences of equal length, where the lengths of the sequences are not a power of 2. We apply our specialized correlation algorithm to analyze data collected in a room-acoustic environment to simultaneously obtain impulse responses between multiple sources and multiple receivers.

Static and dynamic signal processing methods for practical use based on some extension of Gaussian probability in stochastic evaluation of room EM and acoustic environment.

In this paper, we have proposed contrastively static and dynamic type two practical and methodological approaches in order to overcome the complexity of actual electromagnetic (abbr. EM) and sound environments. One is an extended regression analysis of static type intended for employing a standard Gaussian distribution for evaluating mutual relationships between sound and light leaked from a video display terminal (abbr. VDT) in a room, and the other is an establishment of Kalman’s filtering algorithm of dynamic type with help of an equivalence transformation toward the standard Gaussian distribution for estimating a sound absorption coefficient in a reverberation room. The effectiveness of the proposed methods has been also experimentally confirmed by some applications to an actual stochastic evaluation of room lighting and acoustics.

Delayless frequency domain acoustic echo cancellation

The computational complexity of classical time domain gradient-based echo cancellation algorithms might be prohibitively high, due to the very long response of the acoustic transfer functions involved. A reduction in computational complexity can be achieved by using frequency domain or subband algorithms. However, these algorithms introduce an inherent delay in the signal path. The delayed echo has an annoying psychoacoustic effect. Additionally, the delay prevents natural, full-duplex conversation. Moreover, when operated in practical scenarios, using speech signals in actual room acoustic environments, the convergence and tracking properties of the frequency domain algorithms do not compare favorably with those of the NLMS algorithm. This is because the range of values of the convergence constant that support a stable filter is more restrictive for the frequency domain algorithms. In this study we introduce a new algorithm termed delayless frequency domain (DLFD). The DLFD exhibits performance comparable to that of the NLMS algorithm with a computational complexity comparable to that of standard frequency domain algorithms and without the processing delay.

Intelligibility improvements obtained by an enhancement method applied to speech corrupted by noise and reverberation

A series of psychoacoustic experiments are described which attempt to assess the capability of a Multi-Microphone Sub-band Adaptive Signal (MMSBA) processing scheme, for improving the intelligibility of speech corrupted with noise and reverberation. The processing scheme applies the Least Mean Squares (LMS) adaptive algorithm in frequency delimited sub-bands to process speech signals from simulated and real room acoustic environments with various realistic signal to noise ratios (SNR). The processing scheme aims to take advantage of binaural input channels to perform noise cancellation. The two wide-band signals are split into linear or cochlear distributed sub-bands, then processed according to their sub-band signal characteristics. The results of a series of intelligibility tests are presented in which speech and noise data, generated in simulated and real room conditions, was presented to human volunteer subjects at various SNRs, sub-band distributions and sub-band spacings. The results from both simulated and real room acoustical environments show that the MMSBA processing scheme significantly improves both SNR and intelligibility.

混合モデル型の離散レベル実確率システムに対する一ディジタル信号処理法と室内音環境における残響時間推定問題への適用

混合モデル型の離散レベル実確率システムに対する一ディジタル信号処理法と室内音環境における残響時間推定問題への適用

GATS—problems and solutions

A group audio teleconferencing system (GATS) is an audio communications link between groups of people in different locations. Its operation depends in a complex manner on the operation of the system components: room acoustic environments, terminal equipment, transmission facilities, and conference bridges. The main problems encountered are room reverberation, room and circuit noise, chopping of speech by voice switching, and loss contrast (differences in speech levels from various rooms). Solutions to problems of reverberation will be discussed in accompaning papers (Stockbridge et al. and Curtis et al.). This paper discusses solutions to the problem of loss contrast, in particular, use of standard level sound source in alignment and testing of GATS. References will be made to a working example of GATS—the General Service Administration Teleconference System.