Neural networks have been widely applied in direction-of-arrival (DOA) estimation and source tracking systems. In this paper, we introduce a spherical convolutional recurrent neural network that utilizes Deepsphere, a graph-based spherical convolutional neural network, employing the steered response power with phase transform (SRP-PHAT) power maps as input features for real-time robust sound source DOA estimation and tracking applications. The proposed method achieves a performance similar to that of state-of-the-art 3D convolutional neural networks (3D-CNNs) method and reduces the processing time by 88.6%, the parameter count by 85.5%, and the training memory usage by 54.0% respectively. The shallow structure of proposed network demonstrates effectiveness and efficiency.

This study proposes a trainable adaptive window switching (AWS) method and apply it to a deep-neural-network (DNN) for speech enhancement in the modified discrete cosine transform domain. Time-frequency (T-F) mask processing in the short-time Fourier transform (STFT)-domain is a typical speech enhancement method. To recover the target signal precisely, DNN-based short-time frequency transforms have recently been investigated and used instead of the STFT. However, since such a fixed-resolution short-time frequency transform method has a T-F resolution problem based on the uncertainty principle, not only the short-time frequency transform but also the length of the windowing function should be optimized. To overcome this problem, we incorporate AWS into the speech enhancement procedure, and the windowing function of each time-frame is manipulated using a DNN depending on the input signal. We confirmed that the proposed method achieved a higher signal-to-distortion ratio than conventional speech enhancement methods in fixed-resolution frequency domains.

This paper presents an end-to-end deep neural network (DNN)-based source enhancement on the basis of a time-frequency (T-F) mask processing in the modified discrete cosine transform (MDCT)-domain. To retrieve the target signal perfectly in the discrete Fourier transform (DFT)-domain, both amplitude and phase of the spectrum need to be manipulated. However, since it is difficult to deal with complex values by neural network straightforward way, a real-valued T-F mask is commonly estimated and only amplitude spectrum is manipulated. In this study, we use the MDCT instead of the DFT and estimate real-valued T-F masks in the MDCT-domain. The perfect retrieval can be achieved by manipulating only the real-valued MDCT-spectra. To reduce time-domain aliasing arises from manipulating the MDCT spectrum, we build an end-to-end DNN-based source enhancement using T-F mask and train the DNN to minimize an objective function defined in the time-domain. In experiments using several kinds of objective sound quality scores, we observed that the scores were significantly improved.

A method to create multipoles comprising a cluster of focused sources by using a linear loudspeaker array has recently been investigated. Directivities in a listening area were confirmed with examples of primitive multipoles such as dipoles and quadrupoles. This paper describes a method to create a sound source having more complex directivity by using a superposition of multipoles comprising a collection of focused sources. An analytical method is also described with which coefficients can be obtained for each multipole by circular harmonic expansion of a sound field created by a directional sound source and Taylor expansion of the corresponding sound field. Simulation results show that a superposition of multipoles based on analytical conversion introduces desired directivities to the sound sources created in the listening area by a linear loudspeaker array.

We propose a training method for deep neural network (DNN) based source enhancement to increase objective sound quality assessment (OSQA) scores such as the perceptual evaluation of speech quality. In many conventional studies, DNNs have been used as a mapping function to estimate time-frequency masks and trained to minimize an analytically tractable objective function such as the mean squared error (MSE). Since OSQA scores have been used widely for sound-quality evaluation, constructing DNNs to increase OSQA scores would be better than using the minimum MSE to create high-quality output signals. However, since most OSQA scores are not analytically tractable, i.e., they are black boxes, the gradient of the objective function cannot be calculated by simply applying backpropagation. To calculate the gradient of the OSQA-based objective function, we formulated a DNN optimization scheme on the basis of black-box optimization, which is used for training a computer that plays a game. For a black-box-optimization scheme, we adopt the policy gradient method for calculating the gradient on the basis of a sampling algorithm. To simulate output signals using the sampling algorithm, DNNs are used to estimate the probability density function of the output signals that maximize OSQA scores. The OSQA scores are calculated from the simulated output signals, and the DNNs are trained to increase the probability of generating the simulated output signals that achieve high OSQA scores. Through several experiments, we found that OSQA scores significantly increased by applying the proposed method, even though the MSE was not minimized.

We investigated whether a deep neural network (DNN)-based source enhancement function can be self-optimized by reinforcement learning (RL). The use of a DNN is a powerful approach to describing the relationship between two sets of variables and can be useful for source enhancement function design. By training the DNN using a huge amount of training data, sound quality of output signals are improved. However, collecting a huge amount of training data is often difficult in practice. To use limited training data efficiently, we focus on the “self-optimization” of DNN-based source enhancement function in which RL is commonly utilized in the development of game playing computers. As a reward for RL, quantitative metrics that reflect a human's perceptual score (perceptual score), e.g., perceptual evaluation methods for audio source separation (PEASS), are utilized. To investigate whether the sound quality is improved by RL-based source enhancement, subjective tests were conducted. It was confirmed that the output sound quality of the RL-based source enhancement function improved as the number of iterations was increased and finally outperformed the conventional method.

We propose a novel close-talking spherical microphone array that uses the residual signal between the observed sound pressure and the interpolated sound pressure at the center of the spherical array. The interpolated sound is obtained from the sound pressures observed on the surface of a sphere on the basis of the spherical harmonic expansion, assuming that the sound originates from the outside of the array. If the sound source is close to the spherical array, the array cannot express the spherical wave correctly because the number of microphones is limited. As a result, the residual signal increases. This method is a modified form of the conventional method, which interpolates the sound pressure by using the Kirchhoff integral equation. In contrast with the conventional method, we interpolate the sound at the center of the sphere by using only the average value of the sound pressures on the spherical array surface. The computer simulations were conducted using a 12-element spherical microphone array with radius of 5 cm. These results showed that the performances of both methods were almost equivalent, although the proposed method used half the number of microphones as the conventional method.

For sound field reproduction that includes height (with-height reproduction), it is more efficient to record and reproduce the sound field with lower resolution in elevation than in azimuth due to the spatial abilities of human auditory perception. We propose a sound field reproduction method using horizontally arranged cylindrical arrays of microphones and loudspeakers, which is based on the wave field reconstruction (WFR) filter. With the use of cylindrical array configurations, it is possible to reproduce sound waves arriving from upper and lower directions with a smaller number of array elements at angular positions. The WFR filter is analytically derived in the cylindrical harmonic domain and allows direct transformation from the received signals of the microphones into the driving signals of the loudspeakers. A model in which microphones are mounted on a rigid cylindrical baffle is introduced to stabilize the WFR filter. Numerical simulation results indicated that the reproduction accuracy in the neighboring region along the central axis of the cylindrical array was better preserved when using the proposed method than when the method with planar arrays was used.

For with-height reproduction of a sound field, an efficient technique is to record and reproduce the sound field at lower resolution at an elevation angle than that at a horizontal angle based on auditory perception. To achieve this, developing a method using multiple horizontal linear arrays of microphones and loudspeakers is necessary. We propose a sound field reproduction method for a cylindrical array configuration which is based on a wave field reconstruction (WFR) filter analytically derived in the helical wave spectrum domain. The filter is stabilized by introducing a model in which microphones are mounted on a rigid cylindrical baffle. Numerical simulation results indicated that the reproduction accuracy of the proposed method was well preserved in the near-field region along the central axis of the cylinder even when the number of elements at angular position is small.

In real-time sound field transmission systems, the driving signals of a loudspeaker array should be obtained using only the received signals of a microphone array. For efficient transformation of signals of planar or linear arrays, we previously proposed a method of applying a transform filter in the spatio-temporal frequency domain; this filter is defined as the wave field reconstruction (WFR) filter. In linear array configurations, the major artifact of this method is an increase in reverberation time and a decrease in the direct-to-reverberant energy ratio (DRR) because reflections from above and below the microphone array can not be distinguished. We propose a method combining circular harmonics beamforming with the WFR filter as a preprocess in order to match the reproduced DRR to the original one at the time the direct sound wave is properly reproduced. Simulation results indicated that the DRR reproduced using the proposed method was much closer to that reproduced by the method without beamforming.

A method for separating underdetermined sound sources based on a novel power spectral density (PSD) estimation is proposed. The method enables up toM(M-1)+1 sources to be separated when we use a microphone array ofMsensors and a Wiener post-filter calculated by the estimated PSDs. The PSD of a beamformer's output is modelled by a mixture of source PSDs multiplied by the beamformer's directivity gain in the particular angle where each source is located. Based on this model, the PSD of each sound source is estimated from the PSD of multiple fixed beamformers' outputs using the difference in the combination of directivity gains. Simulation results proved that the proposed method effectively separated up toM(M-1)+1 sound sources if the fixed beamformers were appropriately selected. Experiments were also conducted in a reverberant chamber to ensure the proposed method was also effective in practical use.

For transmission of a physical sound field in a large area, it is necessary to transform received signals of a microphone array into driving signals of a loudspeaker array to reproduce the sound field. We propose a method for transforming these signals by using planar or linear arrays of microphones and loudspeakers. A continuous transform equation is analytically derived based on the physical equation of wave propagation in the spatio-temporal frequency domain. By introducing spatial sampling, the uniquely determined transform filter, called a wave field reconstruction filter (WFR filter), is derived. Numerical simulations show that the WFR filter can achieve the same performance as that obtained using the conventional least squares (LS) method. However, since the proposed WFR filter is represented as a spatial convolution, it has many advantages in filter design, filter size, computational cost, and filter stability over the transform filter designed by the LS method.

We generalized our previously proposed diffused sensing for a microphone array design to achieve sharp directive beamforming to enable various filter design methods to be applied. In the conventional microphone array, various filter design methods have been studied to narrow the directivity beam width. However, it is difficult to minimize the power of interference sources in the beamforming output (output interference power) over a broad frequency range since the cross-correlation between transfer functions from sound sources to microphones increases in some frequencies. With the diffused sensing, the cross-correlation is minimized by physically varying the transfer functions. We investigated how a microphone array should be designed in order to minimize the cross-correlation between transfer functions and found that placing the array in a diffuse acoustic field produces optimum results. Because the transfer functions are known a priori, this finding makes it possible to narrow the directivity beam width over a broad frequency range. This observation can be practically achieved by placing microphones inside a reflective enclosure, part of which is open to let sound waves enter. We conducted experiments using 24 microphones and confirmed that the output interference power was reduced over a broad frequency range and the beam width was narrowed by using the diffused sensing.

We present a localized sound reproduction method in which sound energy is high in specified target areas and low everywhere else. Such a system needs to reproduce a sound which attenuates faster than a point source. Our study focuses on an acoustic evanescent wave that has faster distance attenuation. The evanescent wave can be generated when a wave transmitted at subsonic speed between two media. Previously, we used a linear loudspeaker array to reproduce the evanescent wave. Nevertheless, since this method requires a line source of infinite length, a large number of loudspeakers are needed to realize sufficiently fast distance attenuation. To reduce the number of loudspeakers, we propose an evanescent wave reproduction method by using a circular loudspeaker array. Computer simulation results show that our method achieves faster distance attenuation than a point source and reduces the number of loudspeakers to less than that of the linear loudspeaker array.

For real-time sound field transmission systems from a far-end to a near-end, the driving signals of a loudspeaker array at the near-end need to be calculated by using only received signals obtained by a microphone array at the far-end. Additionally, having the capability to control the location of the sound field to be reproduced in order to adjust it to the visual images is advantageous. The goal of this study was to develop a method to transform received signals of a microphone array into driving signals of a loudspeaker array in order to reproduce arbitrary shifted sound fields. We analytically derive a transform filter in the spatio-temporal frequency domain. The location of the sound field to be reproduced is controllable only by phase-shift of the transform filter. The proposed method was found to be computationally efficient compared to the conventional method based on a least squares algorithm, and numerical simulation results indicated that reproduction accuracies were almost the same in both methods.

A spatial sensitivity control method for segregating the sound sources in the same direction by using an acoustic reflector is proposed. Our goal is to clearly pick up the target source at an arbitrary position using a microphone array. Though many methods have been studied for spatial sensitivity control, it is difficult to robustly suppress the power of noise sources in the same direction of the target source in a room. To overcome this problem, we attach a reflector to a microphone array to capture the reflected sounds whose characteristics vary depending on the distance from the array to the source. Assuming that the acoustical properties of the reflector are known e.g., measuring the transfer functions, those reflected sounds can be used as effective clues for segregating the sound sources in the same direction. With the proposed method, a filter for minimizing the output noise power is derived by taking into consideration of the acoustic properties of the reflector. Experiments were conducted in an actual room by using 96 microphones and a large reflector. We confirmed that the spatial sensitivities for segregating the target source at an arbitrary position from noise sources can be achieved by using the proposed method.

We propose a method for achieving sharp directivity by sensing signals in a diffuse acoustic field. Directivity control based on a beamforming method has been studied to make it possible to extract the waveform and location of an identified target source even if there are many noise sources. Sharp directivity can be achieved by minimizing the output noise power of a beamforming filter. However, it is difficult to minimize the output noise power over a broad frequency ranges. Our approach for minimizing the output noise power is to control the spatial properties of the transfer functions and the spatial correlation matrix, by using a reflector that surrounds a microphone array. We investigated the relationships between the output noise power and the structure of the spatial correlation matrix and found that it was possible to minimize the output noise power by sensing diffuse acoustic signals and by designing filters taking the diffuseness of the acoustic field into consideration. In experiments, we observed diffusely reflected signals by placing a truncated-octahedral reflector near a spherical microphone array. We designed filters by using measured transfer functions and confirmed that the proposed method was effective for reducing the output noise power and forming a sharp directivity beamforming filter.

It has been possible to reproduce point sound sources between listeners and a loudspeaker array by using the focused-source method. However, this method requires physical parameters of the sound sources to be reproduced, such as source positions, directions, and original signals. This fact makes it difficult to apply the method to real-time reproduction systems because decomposing received signals into such parameters is not a trivial task. This paper proposes a method for recreating virtual sound sources in front of a planar or linear loudspeaker array. The method is based on wave field synthesis but extended to include the inverse wave propagator often used in acoustical holography. Virtual sound sources can be placed between listeners and a loudspeaker array even when the received signals of a microphone array equally aligned with the loudspeaker array are only known. Numerical simulation results are presented to compare the proposed and focused-source methods. A system was constructed consisting of linear microphone and loudspeaker arrays and measurement experiments were conducted in an anechoic room. When comparing the sound field reproduced using the proposed method with that using the focused-source method, it was found that the proposed method could reproduce the sound field at almost the same accuracy.

We propose a posterior frequency-domain multi-channel residual echo canceling method. As the number of reproduction channels of an echo canceller increases, its adaptive filter shows a slower convergence of the filter coefficient error, that is, large echo path mismatch. In this situation, the change in the human talker in a remote site produces an abrupt residual echo more frequently. The proposed method immediately cancels this abrupt residual echo by estimating it from loudspeaker-reproduced signals, adaptive filter output, and echo replica with a much shorter frame length compared with adaptive filter length. By taking the echo replica into account, this frequency-domain method can improve residual-echo estimates. Another feature of the proposed method is controlling the residual echo estimates according to the confidence interval of these estimates.

We investigated the multipole loudspeaker array based on spherical harmonic expansion in the Cartesian coordinate system. The arrangement of the multipoles with the least number of loudspeaker units has been studied for 2nd order of spherical harmonic expansion. The directivity of the multipole array does not achieve ideal directivity of the spherical functions due to the asymmetrical arrangement. The multipole array was also applied to an end-fire array because a multipole array uses less loudspeakers than the spherical array if the look-direction is fixed. We compared the performance of the conventional least-squares design method and multipole array method for an end-fire array. The least squares method had a higher directivity due to the large number of variable coefficients. Nevertheless, the multipole array based on spherical harmonic expansion has advantages of easy setup in the Cartesian coordinate system, analytical directivity, and small number of filter coefficients.

We present a method for estimating the direct-to-reverberant energy ratio (DRR) that uses a direct and reverberant sound spatial correlation matrix model (Hereafter referred to as the spatial correlation model). This model expresses the spatial correlation matrix of an array input signal as two spatial correlation matrices, one for direct sound and one for reverberation. The direct sound propagates from the direction of the sound source but the reverberation arrives from every direction uniformly. The DRR is calculated from the power spectra of the direct sound and reverberation that are estimated from the spatial correlation matrix of the measured signal using the spatial correlation model. The results of experiment and simulation confirm that the proposed method gives mostly correct DRR estimates unless the sound source is far from the microphone array, in which circumstance the direct sound picked up by the microphone array is very small. The method was also evaluated using various scales in simulated and actual acoustical environments, and its limitations revealed. We estimated the sound source distance using a small microphone array, which is an example of application of the proposed DRR estimation method.

A new approach for estimating the direct-to-reverberant energy ratio (DRR) using a microphone array is proposed. The method is based on a model of a spatial correlation matrix that segregates direct sound and reverberation. It estimates DRR from the power spectra of both components, which are derived from the correlation matrix of the observed signal. In experiments performed in simulated and actual reverberant environments, the proposed method mostly succeeded in estimating DRR accurately. We also present speech enhancement using binary masking as an example of an application of the estimated DRR. By utilization of the DRR as a factor to discriminate the distances of speakers, separation of speech signals whose sources were located in the same direction but at different distances was achieved.

We propose a method for estimating the sound source orientation by using the reflection sounds. The sound source orientation is important spatial information for promoting communication using teleconference systems. We assume that the observed signals captured using several microphones in a reverberant room are used for estimating the sound source orientation. Since the power of each reflection sound depends on the sound source orientation, the transfer functions between a sound source and multiple microphones are varied corresponding to the sound source orientation. We found that the eigenspace of spatial correlation constructed from the observed signals has a characteristic shape corresponding to the sound source orientation. We also proposed an efficient method for estimating the sound source orientation by matching the eigenspace of observed signals with pre-learned eigenspace models for every sound source orientation. In numerical experiments, we obtained about 80% accuracy. We confirmed the effectiveness of the proposed method.

We propose a novel step-size control method that treats the echo path imbalance problem in stereo echo cancellation. The imbalance between two echo paths in magnitude slows down the convergence rate of the adaptive filter for stereo echo cancellation. To speed up the convergence rate, the proposed method extracts the information about filter-coefficient errors of each unknown echo path, and controls the step-size of each channel accordingly.

A fast adaptive algorithm is required for stereo echo cancellation because the strong interchannel cross-correlation of stereo received signals leads to an ill-conditioned normal equation to be solved by the adaptive filter. An adaptive algorithm for this task should also be robust against disturbances such as near-end speech and near-end noise. We propose a coherence-based method of step-size control that provides robustness in stereo echo cancellation. Computer simulation demonstrates that the method is robust against near-end speech and noise, and is capable of quickly tracking changes in echo paths.

Highly cross-correlated input signals create the problem of slow convergence of misalignment in stereo echo cancellation even after undergoing non-linear preprocessing. We propose a new frequency-domain adaptive algorithm that improves the convergence rate by increasing the contribution of non-linearity in the adjustment vector. Computer simulation showed that it is effective when the non-linearity gain is small.

A local active noise control (ANC) system with a virtual microphone based on the common-acoustical-pole and residue (CAPR) model is proposed.

With broadband networks spreading, high-presence acoustic communication like stereophonic full-duplex communication is desired. When loudspeakers and microphones are used for handsfree communication, a stereophonic acoustic echo canceller (SAEC) is required to avoid howling and echoes. However, when there is a large power difference between stereo signals received from the far-end, the performance of the acoustic echo canceller is degraded; that is the convergence speed of the coefficient error is low. In this paper, to overcome this problem, we propose a new adaptive algorithm for the SAEC that uses the adjustment vector. Each channel received a signal normalized by the root of their squared sum. Computer simulation shows that the proposed algorithm improves the convergence speed for stereo signals with a power difference.

Sound localization is important to make conversation easy between local and remote sites in a teleconference. This requires a multi-channel sound system having a multi-channel acoustic echo canceller (MAEC). The appropriate number of channels is determined from a trade-off between high presence and MAEC performance, so it is not possible to increase the channel number by much. We propose a channel-number-compressed MAEC to provide teleconferencing systems that exhibit high presence. The channel number of the MAEC inputs is compressed and that of its outputs is expanded.

Stereo echo cancellation has been achieved and used in daily teleconferencing. To overcome the non-uniqueness problem, a stereo shaker is introduced in eight frequency bands and adjusted so as to be inaudible and not affect stereo perception. A due-filter control system including a continually running adaptive filter and a fixed filter is used for double-talk control. A second-order stereo projection algorithm is used in the adaptive filter. A stereo voice switch is also included. This stereo echo canceller was tested in two-way conversation in a conference room, and the strength of the stereo shaker was subjectively adjusted. A misalignment of 20 dB was obtained in the teleconferencing environment, and changing the talker's position in the transmission room did not affect the cancellation. This echo canceller is now used daily in a high-presence teleconferencing system and has been demonstrated to more than 300 attendees.

A method is proposed for modeling a room transfer function (RTF) by using common acoustical poles and their residues. The common acoustical poles correspond to the resonance frequencies (eigenfrequencies) of the room, so they are independent of the source and receiver positions. The residues correspond to the eigenfunctions of the room. Therefore, the residue, which is a function of the source and receiver positions, can be expressed using simple analytical functions for rooms with a simple geometry such as rectangular. That is, the proposed model can describe RTF variations using simple residue functions. Based on the proposed common-acoustical-pole and residue model, methods are also proposed for spatially interpolating and extrapolating RTFs. Because the common acoustical poles are invariant in a given room, the interpolation or extrapolation of RTFs is reformulated as a problem of interpolating or extrapolating residue values. The experimental results for a rectangular room, in which the residue values are interpolated or extrapolated by using a cosine function or a linear prediction method, demonstrate that unknown RTFs can be well estimated at low frequencies from known (measured) RTFs by using the proposed methods.

Use of a common-acoustical-pole and zero model is proposed for modeling head-related transfer functions (HRTFs) for various directions of sound incidence. The HRTFs are expressed using the common acoustical poles, which do not depend on the source directions, and the zeros, which do. The common acoustical poles are estimated as they are common to HRTFs for various source directions; the estimated values of the poles agree well with the resonance frequencies of the ear canal. Because this model uses only the zeros to express the HRTF variations due to changes in source direction, it requires fewer parameters (the order of the zeros) that depend on the source direction than do the conventional all-zero or pole/zero models. Furthermore, the proposed model can extract the zeros that are missed in the conventional models because of pole-zero cancellation. As a result, the directional dependence of the zeros can be traced well. Analysis of the zeros for HRTFs on the horizontal plane showed that the nonminimum-phase zero variation was well formulated using a simple pinna-reflection model. The common-acoustical-pole and zero (CAPZ) model is thus effective for modeling and analyzing HRTF's.

A new configuration for a stereo echo canceller with nonlinear pre-processing is proposed. The pre-processor which adds uncorrelated components to the original received stereo signals improves the adaptive filter convergence even in the conventional configuration. However, because of the inaudibility restriction, the preprocessed signals still include a large amount of the original stereo signals which are often highly cross-correlated. Therefore, the improvement is limited. To overcome this, our new stereo echo canceller includes exclusive adaptive filters whose inputs are the uncorrelated signals generated in the pre-processor. These exclusive adaptive filters converge to true solutions without suffering from cross-correlation between the original stereo signals. This is demonstrated through computer simulation results.

This paper proposes a new subband stereo echo canceller that converges to the true echo path impulse response much faster than conventional stereo echo cancellers. Since signals are bandlimited and downsampled in the subband structure, the time interval between the subband signals become longer, so the variation of the crosscorrelation between the stereo input signals becomes large. Consequently, convergence to the true solution is improved. Furthermore, the projection algorithm, or affine projection algorithm, is applied to further speed up the convergence. Computer simulations using stereo signals recorded in a conference room demonstrate that this method significantly improves convergence speed and almost solves the problem of stereo echo cancellation with low computational load.

A multiple-point equalization filter using the common acoustical poles of room transfer functions is proposed. The common acoustical poles correspond to the resonance frequencies, which are independent of source and receiver positions. They are estimated as common autoregressive (AR) coefficients from multiple room transfer functions. The equalization is achieved with a finite impulse response (FIR) filter, which has the inverse characteristics of the common acoustical pole function. Although the proposed filter cannot recover the frequency response dips of the multiple room transfer functions, it can suppress their common peaks due to resonance; it is also less sensitive to changes in receiver position. Evaluation of the proposed equalization filter using measured room transfer functions shows that it can reduce the deviations in the frequency characteristics of multiple room transfer functions better than a conventional multiple-point inverse filter. Experiments show that the proposed filter enables 1-5 dB additional amplifier gain in a public address system without acoustic feedback at multiple receiver positions. Furthermore, the proposed filter reduces the reflected sound in room impulse responses without the pre-echo that occurs with a multiple-point inverse filter. A multiple-point equalization filter using common acoustical poles can thus equalize multiple room transfer functions by suppressing their common peaks.

This paper proposes a new subband echo canceller that almost fully whitens the received input by using the low-order projection algorithm, or the affine projection algorithm. Since the projection algorithm can fully whiten the received input of a small-tap adaptive filter with a relatively small projection order, the proposed subband projection echo canceller achieves nearly maximum convergence with a small projection order. By reflecting the frequency characteristics of the speech and of the echo path, it allows a different projection order to be chosen in different subbands. This gives the proposed method optimum cost performance, which is promising for implementation.

A new model for a room transfer function (RTF) by using common acoustical poles that correspond to resonance properties of a room is proposed. These poles are estimated as the common values of many RTF's corresponding to different source and receiver positions. Since there is one-to-one correspondence between poles and AR coefficients, these poles are calculated as common AR coefficients by two methods: (i) using the least squares method, assuming all the given multiple RTF's have the same AR coefficients and (ii) averaging each set of AR coefficients estimated from each RTF. The estimated poles agree well with the theoretical poles when estimated with the same order as the theoretical pole order. When estimated with a lower order than the theoretical pole order, the estimated poles correspond to the major resonance frequencies, which have high Q factors. Using the estimated common AR coefficients, the proposed method models the RTF's with different MA coefficients. This model is called the common-acoustical-pole and zero (CAPZ) model, and it requires far fewer variable parameters to represent RTF's than the conventional all-zero or pole/zero model. This model was used for an acoustic echo canceller at low frequencies, as one example. The acoustic echo canceller based on the proposed model requires half the variable parameters and converges 1.5 times faster than one based on the all-zero model, confirming the efficiency of the proposed model.>

A method for modeling a room transfer function (RTF) by using estimated common acoustical poles that correspond to resonance properties of a room is proposed. These poles are estimated as common values of the multiple RTFs corresponding to different source and receiver positions. This common-acoustical-pole and zero (CAPZ) model requires far fewer variable parameters to represent RTFs than conventional all-zero or pole/zero models. This model was applied to an acoustic echo canceller and to head-related transfer functions. At low frequencies, the acoustic echo canceller based on this model converges 1.5 times faster than the one based on the all-zero model. Head-related transfer functions that have resonance characteristics of the external ear are also successfully modeled by the proposed model.>