Abstract: The scale of an apparatus that reproduces a sound field is reduced. A speaker array 1 includes a plurality of speakers arranged at intersections of a first plurality of virtual lines arranged parallel to each other at equal intervals and a second plurality of virtual lines that are perpendicular to the first plurality of virtual lines and are arranged parallel to each other at equal intervals, wherein multipoles of a given order are superimposed using the plurality of speakers to realize wave field synthesis.

Abstract: The scale of an apparatus that reproduces a sound field is reduced. A speaker array 1 includes a plurality of speakers arranged at intersections of a first plurality of virtual lines arranged parallel to each other at equal intervals and a second plurality of virtual lines that are perpendicular to the first plurality of virtual lines and are arranged parallel to each other at equal intervals, wherein multipoles of a given order are superimposed using the plurality of speakers to realize wave field synthesis.

Abstract: Provided is a sound image reproduction device, sound image reproduction method, and sound image reproduction program that can support monaural sound sources and is capable of imparting directivity to virtual sound sources in a space.

Abstract: An acoustic signal processing device 1 includes: a focal point position determination unit 12 that obtains a plurality of sets of initial focal point coordinates, coordinates of the virtual sound source, and a direction of directivity thereof, and for a pair of sets of initial focal point coordinates with different polarities among the plurality of sets of initial focal point coordinates, multiplies the sets of initial focal point coordinates by a rotation matrix based on the coordinates of the virtual sound source to thereby determine sets of focal point coordinates, the rotation matrix being specified from the direction of the directivity; a circular harmonic coefficient conversion unit 13 that calculates weights to be applied to multipoles including the sets of focal point coordinates from a circular harmonic coefficient; a filter coefficient computation unit 14 that, for each of the speakers in the speaker array, computes a weighted driving function to be applied to the speaker from the sets of focal poin

Abstract: Provided is a sound image reproduction device, sound image reproduction method, and sound image reproduction program that can support monaural sound sources and is capable of imparting directivity to virtual sound sources in a space.

Abstract: An acoustic signal processing device 1 includes: a focal point position determination unit 12 that obtains a plurality of sets of initial focal point coordinates, coordinates of the virtual sound source, and a direction of directivity thereof, and for a pair of sets of initial focal point coordinates with different polarities among the plurality of sets of initial focal point coordinates, multiplies the sets of initial focal point coordinates by a rotation matrix based on the coordinates of the virtual sound source to thereby determine sets of focal point coordinates, the rotation matrix being specified from the direction of the directivity; a circular harmonic coefficient conversion unit 13 that calculates weights to be applied to multipoles including the sets of focal point coordinates from a circular harmonic coefficient; a filter coefficient computation unit 14 that, for each of the speakers in the speaker array, computes a weighted driving function to be applied to the speaker from the sets of focal poin

Abstract: A sound enhancement technique that uses transfer functions ai,g of sounds that come from each of one or more positions/directions that are assumed to be sound sources arriving at each microphone to obtain a filter for a position that is a target of sound enhancement, where i denotes a direction and g denotes a distance for identifying each of the positions. Each of the transfer functions ai,g is represented by sum of a transmission characteristic of a direct sound that directly arrives from the position determined by the direction i and the distance g and a transmission characteristic of one or more reflected sounds produced by reflection of the direct sound off an reflective object. A filter that corresponds to the position that is the target of sound enhancement is applied to frequency-domain signals transformed from M picked-up sounds picked up with M microphones to obtain a frequency-domain output signal.

Abstract: A sound enhancement technique that uses transfer functions ai,g of sounds that come from each of one or more positions/directions that are assumed to be sound sources arriving at each microphone to obtain a filter for a position that is a target of sound enhancement, where i denotes a direction and g denotes a distance for identifying each of the positions. Each of the transfer functions ai,g is represented by sum of a transmission characteristic of a direct sound that directly arrives from the position determined by the direction i and the distance g and a transmission characteristic of one or more reflected sounds produced by reflection of the direct sound off an reflective object. A filter that corresponds to the position that is the target of sound enhancement is applied to frequency-domain signals transformed from M picked-up sounds picked up with M microphones to obtain a frequency-domain output signal.

Abstract: Even if received signals are highly cross-correlated, echoes can be effectively cancelled and no psychoacoustical problems arise. A received signal xi(k) (where i=1, 2, . . . , N) and an additive signal ai(k) are added together, and the added output is used to drive a speaker i and input into an echo cancellation filter 405i. The received signal xi(k) and the additive signal ai(k) are input into adaptive filters 401i and 402i, respectively. The difference between the sum of the outputs from all the filters 401i and all the filters 402i and an echo ym(k) is detected as an error em(k). The coefficients of all the filters 401i and 402i are updated to reduce the error em(k). When the error em(k) is made sufficiently small, the coefficients of the filters 402i are transferred to the filters 405i. The sum of the outputs from all the filters 405i is detected as an echo replica, and the difference between the echo replica and the echo ym(k) is output.

Abstract: In a subband echo cancellation for a multichannel teleconference, received signals x1(k), x2(k), . . . , xI(k) of each channel are divided into N subband signals, an echo y(k) picked up by a microphone 16j after propagation over an echo path is divided into N subband signals y0(k), . . . ,yN−1(k), and vectors each composed of a time sequence of subband received signals x1(k), . . . , xI(k) are combined for each corresponding subband. The combined vector and an echo cancellation error signal in the corresponding subband are input into an estimation part 19n, wherein a cross-correlation variation component is extracted. The extracted component is used as an adjustment vector to iteratively adjust the impulse response of an estimated echo path. The combined vector is applied to an estimated echo path 18n formed by the adjusted value to obtain an echo replica. An echo cancellation error signal en(k) is calculated from the echo replica and a subband echo yn(k).

Abstract: A received signal is output to an echo path and, at the same time, it is divided into a plurality of subbands to generate subband received signals, which are applied to estimated echo paths in the respective subbands to produce echo replicas. The echo having propagated over the echo path is divided into a plurality of subbands to generate subband echoes, from which the corresponding echo replicas are subtracted to produce misalignment signals. Based on the subband received signal in each subband and the misalignment signal corresponding thereto, a coefficient to be provided to each estimated echo path is adjusted by a projection or ES projection algorithm.

Abstract: In a subband acoustic echo canceller which generates an echo replica from a subband received signal x.sub.k (m) by an estimated echo path in each subband, subtracts the echo replica from a subband echo signal y.sub.k (m) by a subtractor to generate a subband error signal e.sub.k (m) and uses an adaptive algorithm in an echo path estimation part to estimate the transfer function of the estimated echo path from the subband error signal e.sub.k (m) and the subband received signal x.sub.k (m) so that the subband error signal e.sub.k (m) approaches zero, the stop-band attenuation of each band-pass filter of a received signal subband analysis part for generating the subband received signal x.sub.k (m) is set to be smaller than the stop-band attenuation of each band-pass filter of an echo subband analysis part for generating the subband echo signal Y.sub.k (m) to thereby flatten the frequency characteristics of the subband received signals relative to the subband echo signals.

Abstract: In a subband acoustic echo canceller, FG/BG filters are provided in M ones of N subbands into which the received signal is divided, and adaptive filters are provided in the other remaining subbands. In the respective FG/BG filters, during the detection of a non-double-talk state their transfer logic parts output state signals GD-j, GD-k, . . . and their adaptive operation control parts each apply an adaptation condition signal ADP to the adaptive filter in each of the above-mentioned other remaining subbands when a predetermined number or more of the FG/BG filters output the state signals GD-j, GD-k, . . . The adaptive filter updates the subband estimated echo path coefficient only when it is supplied with the signal ADP.

Abstract: In an adaptive estimation of an acoustic transfer function of an unknown system, a forward linear prediction coefficient vector a(k) of an input signal x(k), the sum of forward a posteriori prediction-error squares F(k), a backward linear prediction coefficient vector b(k) of the input signal x(k) and the sum of backward a posteriori prediction-error squares B(k) are computed. Letting a step size and a pre-filter deriving coefficient vector be represented by .mu.

Abstract: In an echo cancelling method of a p-order fast projection algorithm which subtracts an estimated echo signal y(k) from a microphone output signal u(k) to obtain an error signal e(k), adaptively calculates a pre-filter coefficient .beta.(k) from the auto-correlation of a received speech signal x(k) and the error signal, generating an intermediate variable z(k) updated by a coefficient s(k) obtained by smoothing the pre-filter coefficient, convolutes the received speech signal x(k) and the intermediate variable z(k), calculates the inner product of the auto-correlation of the received speech signal and the smoothed pre-filter coefficient s(k) and adding the inner product and the convoluted output to obtain the estimated echo signal, the magnitudes of the received speech signal x(k) and the error signal e(k) are compared and when the result of comparison satisfies a predetermined condition, a reset signal is generated to set the pre-filter coefficient .beta.

Abstract: A received input signal and an echo signal resulting from the passage of the received input signal through an echo path are both analyzed or divided into a plurality of common subbands. The received input signal in each subband is supplied to an estimated echo path provided in the subband, by which it is rendered into an echo replica signal. The echo replica signal is subtracted, by a subtractor provided in each subband, from the echo signal in the same subband as the echo replica signal to obtain a residual echo signal. The residual echo signals in the respective subbands are synthesized into a full-band residual echo signal. The estimated echo path in each subband is formed by a digital FIR filter and its filter coefficients are calculated by a coefficient calculation part in the subband, based on the received input signal, the residual echo signal and a step size matrix. The filter coefficients are iteratively updated so that the residual echo signal in each subband may be minimized.

Abstract: A plurality of acoustic transfer functions for a plurality of sets of different positions of a loudspeaker and a microphone in an acoustic system are measured by an acoustic transfer function measuring part. The plurality of measured acoustic transfer functions are used to estimate poles of the acoustic system by a pole estimation part, and a fixed AR filter is provided with the estimated poles as fixed values. A variable MA filter is connected in series to the fixed AR filter and the acoustic transfer function of the acoustic system is simulated by the two filters. The filter coefficients of the variable MA filter are modified with a change in the acoustic transfer function of the acoustic system.