Effects Of Room Reverberation Research Articles

A robust denoising process for spatial room impulse responses with diffuse reverberation tails.

Spatial room impulse responses (SRIRs) measured using spherical microphone arrays are seeing increasingly widespread use in reproducing room reverberation effects on three-dimensional surround sound systems (e.g., higher-order ambisonics) through multi-channel SRIR convolution. However, such measured impulse responses inevitably present a non-negligible noise floor, which may lead to a perceptible "infinite reverberation effect" when convolved with an input sound. Furthermore, individual sensor noise and momentary measurement artefacts may additionally corrupt the resulting impulse response. This paper presents a robust SRIR denoising procedure applicable to impulse responses with diffuse late reverberation tails, which can be modeled by a stochastic process. In such cases, the non-decaying frequency-dependent noise floor may be replaced by a synthesized incoherent tail parameterized by the SRIR's energy decay envelope. It is shown that performing such tail re-synthesis in the spherical harmonic domain, using an independent zero-mean Gaussian noise for each component, preserves both the reverberation tail's frequency-dependent decay as well as its spatial coherence properties. The proposed process is then evaluated through its application to SRIRs measured in real-world conditions, and finally some aspects of performance and consistency verification are discussed.

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.

Detection of reflections in different room acoustical conditions using generalized cross correlation phase transform

A framework for localization of the received source emission was implemented in enclosures based on generalized cross correlation (GCC) weighting with phase transform (PHAT) function. It was to achieve the goal of capturing meaningful sounds including speech in actual environments. The effects of room reverberation and ambient noise on the performance of the system were investigated with a linear microphone array. Experiments were performed in different environments with different RT and background noise levels. The sound sources are consisting of pink noise and speech sources were used to observe the relationship between localization accuracy and room acoustical parameters. Results for evaluation indicated that the location of the reflections was noticeable for accurately detection of the sources in reverberant and noisy environments.

A Statistically Robust Approach to Acoustic Impulse Response Shaping

Acoustic impulse response shaping is a prefiltering technique to reduce the perceptible effects of room reverberation. The room impulse responses to be shaped must first be measured, and these measurements can contain errors. Furthermore, room responses vary with changes in temperature and humidity and also with changes in measurement position. This letter presents a method for enhancing the robustness of the shaping procedure so that the effects of these errors and changes are minimized. The method uses a stochastic model of the channel variations to explicitly limit the probability of large deviations from the desired performance. The method is evaluated on realistic channel perturbations and the resulting shaped responses are shown to comply with the robustness specification.

Features for Masking-Based Monaural Speech Separation in Reverberant Conditions

Monaural speech separation is a fundamental problem in speech and signal processing. This problem can be approached from a supervised learning perspective by predicting an ideal time–frequency mask from features of noisy speech. In reverberant conditions at low signal-to-noise ratios (SNRs), accurate mask prediction is challenging and can benefit from effective features. In this paper, we investigate an extensive set of acoustic–phonetic features extracted in adverse conditions. Deep neural networks are used as the learning machine, and separation performance is evaluated using standard objective speech intelligibility metrics. Separation performance is systematically evaluated in both nonspeech and speech interference, in a variety of SNRs, reverberation times, and direct-to-reverberant energy ratios. Considerable performance improvement is observed by using contextual information, likely due to temporal effects of room reverberation. In addition, we construct feature combination sets using a sequential floating forward selection algorithm, and combined features outperform individual ones. We also find that optimal feature sets in anechoic conditions are different from those in reverberant conditions.

Predicting speech intelligibility based on a correlation metric in the envelope power spectrum domain.

19(7), 2125-2136]. This "hybrid" model, named sEPSMcorr, is shown to account for the effects of stationary and fluctuating additive interferers as well as for the effects of non-linear distortions, such as spectral subtraction, phase jitter, and ideal time frequency segregation (ITFS). The model shows a broader predictive range than both the original mr-sEPSM (which fails in the phase-jitter and ITFS conditions) and STOI (which fails to predict the influence of fluctuating interferers), albeit with lower accuracy than the source models in some individual conditions. Similar to other models that employ a short-term correlation-based back end, including STOI, the proposed model fails to account for the effects of room reverberation on speech intelligibility. Overall, the model might be valuable for evaluating the effects of a large range of interferers and distortions on speech intelligibility, including consequences of hearing impairment and hearing-instrument signal processing.

Evaluation of a spectral subtraction strategy to suppress reverberant energy in cochlear implant devices.

The smearing effects of room reverberation can significantly impair the ability of cochlear implant (CI) listeners to understand speech. To ameliorate the effects of reverberation, current dereverberation algorithms focus on recovering the direct sound from the reverberated signal by inverse filtering the reverberation process. This contribution describes and evaluates a spectral subtraction (SS) strategy capable of suppressing late reflections. Late reflections are the most detrimental to speech intelligibility by CI listeners as reverberation increases. By tackling only the late part of reflections, it is shown that users of CI devices can benefit from the proposed strategy even in highly reverberant rooms. The proposed strategy is also compared against an ideal reverberant (binary) masking approach. Speech intelligibility results indicate that the proposed SS solution is able to suppress additive reverberant energy to a degree comparable to that achieved by an ideal binary mask. The added advantage is that the SS strategy proposed in this work can allow for a potentially real-time implementation in clinical CI processors.

The effects of reverberation on a listener's ability to recognize target sentences in the presence of up to three synchronized masking sentences

Objective: To determine the effects of room reverberation on target sentence recognition in the presence of 0-to-3 synchronous masking sentences. Design: Target and masker sentences were presented through four loudspeakers (±90° and ±45° azimuth; 1m from the listener) in rooms having reverberation times (RT) of 0.2, 0.4, 0.6, and 1.1 s. Study Sample: Four groups of 13 listeners each participated in the study (N = 52). Results: In rooms with RTs of 0.2, 0.4, and 0.6 s, mean speech recognition scores (SRSs) were similar, with scores ranging from 96–100%, 90–95%, 75–80%, and 53–60%, when 0, 1, 2, and 3 competing sentences were present, respectively. However, in the room with a RT = 1.1 s, SRSs deteriorated significantly faster as the number of competing sentences increased; mean scores were 93%, 73%, 26%, and 10%, in the 0, 1, 2, 3, competing sentence condition, respectively. The majority of errors in SRSs (98%) resulted from listeners reporting words presented in masking sentences along with those in target sentences (mixing errors). Conclusions: Results indicate that reverberation has a similar influence on SRSs measured in multi-talker environments, when room reverberation is ≤ 0.6 s. However, SRSs are dramatically reduced in the room with a RT = 1.1 s, even when only one competing talker is present.SumarioObjetivo: Determinar los efectos de la reverberación de un cuarto para el reconocimiento de oraciones blanco, en presencia sincrónica de 0–3 oraciones enmascarantes. Diseño: Las oraciones blanco y las enmascarantes se presentaron por medio de cuatro altoparlantes (+90° y ±45° azimuth; a 1m del oyente) en cuartos que tenían tiempos de reverberación (RT) de 0.2, 0.4, 0.6 y 1.1 seg. Muestra de estudio: Participaron en este estudio, cuatro grupos de 13 oyentes cada uno (N=52). Resultados: En cuartos con RT de 0.2, 0.4 y 0.6 seg, las puntuaciones medias de reconocimiento del habla (SRS) fueron similares, con puntuaciones que variaron de 96–100%, 90–95%, 75–80%, y 53–60%, cuando se presentaron, respectivamente, 0, 1, 2, y 3 oraciones competitivas. Sin embargo, en el cuarto con RT=1.1 seg, las SRS se deterioraron significativamente más rápido, conforme el número de frases competitivas se incrementó; las puntuaciones medias fueron de 93%, 73%, 26%, y 10%, en las condiciones de 0, 1, 2 y 3 oraciones competitivas, respectivamente. La mayoría de los errores en las SRS (98%) resultaron de oyentes que reportaron palabras presentes en las oraciones enmascarantes junto con aquellas de las oraciones blanco. (mezcla de errores). Conclusiones: Los resultados indican que la reverberación tiene una influencia similar en las SRS, medidas en ambientes de hablantes múltiples, cuando la reverberación es ≦0.6 sec. Sin embargo, las SRS se reducen dramáticamente en un cuarto con un RT=1.1 seg, incluso cuando solamente está presente un mensaje competitivo.

A real-time audio system for adjusting the sweet spot to the listener's position

In the present study, a new stereophonic playback system was proposed, where the cross-talk signals would be reasonably cancelled at an arbitrary listener position. The system was composed of two major parts: the listener position tracking part and the sound rendering part. The position of the listener was estimated using acoustic signals from the listener (i.e. voice or hand-clapping signals). A direction of arrival (DOA) algorithm was adopted to estimate the directions of acoustic sources where the room reverberation effects were taken into consideration. A Crosstalk cancellation filter was designed using a free-field model. To determine the maximum tolerable shift of the listener position, a quantitative analysis of the channel separation ratio according to the displacement of the listener position was performed. Prototype hardware was implemented using a microprocessor board, a DSP board, a multi-channel ADC board and an analog frontend. The results showed that the average mean square error between the true direction of a listener and the estimated direction was about 5 degrees. More than 80% of the tested subjects indicated that better stereo images were obtained by the proposed system, compared with the non-processed signals.

Effects of room reverberation time and receiver position on measured binaural room impulse responses.

This research examines the effects of room reverberation and receiver position on the fine detail in measured binaural room impulse responses, which can affect psychoacoustic perception in rooms such as speech segregation. Binaural room impulse responses have been gathered in four spaces with a range of shapes and reverberation times for three different receiver positions (center, 1 m from side wall, and 1 m from back wall) and three different source orientations (0, 45, and 90 deg from receiver). The directional loudspeaker is located approximately 0.5 m in front of the receiver for all conditions. Relationships between room reverberation times and measures impacting source localization such as interaural level differences and frequency‐to‐frequency fluctuations are analyzed. The results will determine if conclusions from previous research in this area conducted in a classroom [B.G. Shinn‐Cunningham, N. Kopco, and T.J. Martin, J. Acoust. Soc. Am. 117, 3100–3115 (2005)] are applicable to spaces with varying shapes and reverberation times.

New techniques for measuring speech privacy and efficiency of sound masking systems

Traditionally, speech privacy has been measured by means of the Articulation Index (AI), transposed to rate privacy rather than intelligibility (PI=1-AI). However, this is an indirect and slightly cumbersome method that usually requires a spreadsheet calculation to yield the Privacy Index rating. Furthermore, the traditional AI approach does not take into account room reverberation effects (though the very much more complex SII does). The paper discusses the potential use of STI and STIPa as direct measures of speech privacy and shows that, potentially, they may be able to offer a superior approach. The benefits and limitations of the methods are highlighted together with the results from a number of case studies. It is concluded that while the method has potential merit, a number of the limiting factors require further research. It particular, it is shown that the index scale at lower levels of intelligibility may need to be revised.

Recovering the quality of speech degraded by reverberations in a room

Speech signals captured by distant microphones in a room are usually reverberated due to wall reflections. Reverberation may seriously deteriorate the signal characteristics, thus damaging the quality of such applications as hands‐free telephony or automatic speech recognition. To eliminate such room reverberation effects, the inverse‐filtering of the room transfer functions (RTFs) between a speaker and the microphones appears to be a very promising approach. When applied to a microphone system, this processing should work blindly because no explicit information on RTFs or source signals is available. The blind design of an inverse‐filter set may be roughly classified into three approaches: (1) calculating an inverse‐filter set of RTF estimates obtained from observed reverberant signals based on the ‘‘subspace method,’’ (2) calculating the filter set from observed signals and replicas of direct‐sound signals using statistical properties inherent in speech signals, and (3) calculating the filter set direct...

Effects of room reverberation on spatial release from masking

When signal and masker sources are separated horizontally in an anechoic space, one of the primary benefits is that the acoustic shadow from the head leads to a substantial improvement in signal-to-noise ratio in the high frequencies at one of the ears, relative to conditions in which signal and noise are presented from a common location. This presentation will examine how the improved signal audibility created by head shadow is affected by room reverberation. Through room-simulation models, the effects of three variables will be examined: degree of horizontal separation, distance between the listener and the signal and masker, and degree of room reverberation. This paper will present an evaluation of how well the effects of these variables can be predicted by generalized room acoustics equations incorporating room volume and surface absorption. The reduction in high-frequency head-shadow effects in reverberant environments, combined with a disruption of cues for low-frequency binaural release from masking, severely reduces the benefits of spatially separating signal and noise. Fortunately, due to the precedence effect, the perceived locations of signal and noise are accurate even in a reverberant space. The intelligibility benefits of perceived differences in signal and masker locations will be discussed. [Work supported by NIDCD.]

Room reverberation effects in hearing aid feedback cancellation

Room reverberation can affect feedback cancellation in hearing aids, with the strength of the effects depending on the acoustical conditions. These effects were studied using a behind the ear (BTE) hearing aid mounted on a dummy head and coupled to the ear canal via an open fitting. The hearing aid impulse response was measured for the dummy head placed at eight closely spaced locations in a typical office. The feedback cancellation in the hearing aid used a set of filter coefficients that were initialized for one location within the room, and then allowed to adapt to the feedback path measured at the same or to a different location. The maximum stable gain for the hearing aid was then estimated without feedback cancellation, for the initial set of feedback cancellation filter coefficients prior to adaptation, and for the feedback cancellation filter after adaptation. A low-order ARMA model combining a fixed set of poles with an adaptive FIR filter is shown to be effective in representing the feedback path exclusive of reverberation. Increasing the adaptive filter length has only a small benefit in improving the feedback cancellation performance due to the inability of the system to model the room reverberation. The mismatch between the modeled and actual feedback paths limits the headroom increase that can be achieved when using feedback cancellation, and varies with the location within the room.

Performance of time-delay estimation in the presence of room reverberation

Synthetic microphone signals generated with the image model technique are used to study the effects of room reverberation on the performance of the maximum likelihood (ML) estimator of the time delay, in which the estimate is obtained by maximizing the cross correlation between filtered versions of the microphone signals. The results underscore the adverse effects of reverberation on the bias, variance and probability of anomaly of the ML estimator. Explanations of these effects are provided.

Audio Technology Used in AT&T's Terminal Equipment

This paper discusses the audio technology of AT&T's terminal equipment. It describes electroacoustic and digital signal processing (DSP) techniques that improve the perceived audio quality and ease of use of telephones and speakerphones. The electroacoustic techniques include directional microphones, close-talking microphones, and loudspeakers of varying designs. The DSP techniques include echo control, speech enhancement, and active noise cancellation. These techniques increase signal-to-noise ratios, expand transmitted bandwidth, and reduce the effects of room reverberation, echo, and other common audio problems.

The performance of adaptive noise cancellation systems in reverberant rooms

The performance of adaptive noise cancellation (ANC) systems in a reverberant room is investigated. Room reverberation effects are modeled through the method of images, incorporating both direct signal and reverberation effects into the room impulse response. The performance of ANC for broadband inputs is examined as a function of reverberation time, noise source location, and the geometry of the room and measurement equipment. Expressions are obtained relating the steady-state minimum mean-squared error and excess mean-squared error to the reverberation time and filter parameters. Nonuniform convergence effects are also investigated and bounds on the eigenvalue disparity are obtained for a given configuration. It is shown that the eigenvalue disparity increases as both the filter length and the distance between noise source and reference receiver increase, approaching a limiting value as the distance from source to reference receiver surpasses the acoustic critical distance.

Effects of Reverberation and Noise on Speech Recognition by Adults with Various Amounts of Sensorineural Hearing Impairment

Speech recognition was assessed under three levels of room reverberation, each in quiet and noise, for subjects with varying amounts of sensorineural hearing impairment. The three acoustic environments were: sound suite, reverberant room with a reverberation time (T) = 0.54 s and reverberant room with T = 1.55 s. Three groups of subjects were utilized: normal hearing, mild sensorineural hearing impairment and moderate-to-severe sensorineural hearing impairment. Speech recognition ability for each of the three groups of subjects significantly differed from the other two groups for each of the three reverberant conditions. The detrimental influences of noise and reverberation increased with the magnitude of hearing impairment. In addition, there was an interaction between noise and room reverberation, where the detrimental effects of room reverberation were compounded by the addition of noise.

Effects of room reverberation and noise on speech discrimination by the elderly.

Speech discrimination performance was measured in sound field for 10 young normal-hearing subjects, 10 elderly 'normal-hearing' subjects with a mean pure-tone average of 9.9 dB HTL, and 10 elderly hearing-impaired subjects with a mean pure-tone average of 48.5 dB HTL. Speech discrimination abilities were assessed in quiet and noise (S/N = +10 dB) in a sound suite and under two levels of reverberation in a reverberant room (RT = 0.59 and 1.56 s). Results indicated that the elderly 'normal-hearing' and young normal-hearing subjects have similar speech discrimination performance in the sound suite for both the quiet and noise conditions. In addition, performance by these two groups was almost identical under both levels of reverberation in quiet. However, when noise was added to the reverberant conditions, performance by the elderly 'normal-hearing' subjects was significantly poorer than that obtained by the young normal-hearing subjects. The elderly hearing-impaired subjects yielded speech discrimination scores that were significantly poorer than the elderly 'normal-hearing' subjects and the young normal-hearing subjects for all of the listening conditions.

Effects of room reverberation upon hearing aid quality judgments.

Hearing-aid-processed speech recorded in reverberant and nonreverberant environments was ranked utilizing quality judgment and speech discrimination tasks. 21 normally hearing and 20 hearing-impaired listeners ranked the hearing-aid-processed speech. Quality judgments in reverberant environments may yield important information with respect to hearing aid selection. In contrast, speech discrimination rankings in a nonreverberant environment appeared to be similar to the speech discrimination rankings obtained in the reverberant condition.