Abstract: A system (100) for processing audio data comprises an extracting unit (101) adapted to extract a transient audio data part (103) from input audio data (102) and a reverberator unit (105) which is coupled to the extracting unit (101) so as to be provided with a transient audio data part (103), wherein the reverberator unit (104) is adapted to generate reverberation separately for the transient audio data part (103).

Abstract: A sound field control apparatus which is capable of adjusting the number of outputs as required. The sound field control apparatus is comprised of a master device, and a plurality of salve devices serially connected in cascade to the master device. In the master device, a picked-up sound signal input section inputs picked-up sound signals from a plurality of microphones. A signal processing section generates reflected sound signals from the input picked-up sound signals. A cascade output terminal outputs the reflected sound signals. In each of the slave devices, a cascade input terminal inputs the generated reflected sound signal, a plurality of signal processing sections correspond to a plurality of speakers and each adjust sound characteristics of the input reflected sound signals and generate and output signals to the corresponding speakers. A cascade output terminal directly outputs the input reflected sound signals.

Abstract: An apparatus and method for artificial reverberation having one or more input summing nodes receiving an input signal and which each feed a first series of delay elements and the output of which is thereafter summed, weighted, and fed back to the input summing nodes. Also, a feedback summing node which further feeds a second delay element which feeds an output with a signal which has multi-dimensional or multi-delay reverberation relative to the input signal. An alternative embodiment further feeds one or more of the first delay element outputs weighted via an output gain to an output summing node which also sums the output of the second delay element and provides a summed output having a multi-dimensional or multi-delay reverberation. The embodiment is easily and economically implemented via conventional integrated electronic circuits, a combination of analog and digital electronics, digital signal processors, microprocessors, micro-controllers, or computer algorithms.

Abstract: An audio processing system comprising processing devices and a processing module. The processing devices generate output signals, and the processing module calculates the output signals from the processing devices and generates an output result signal. Each processing device comprises a control module, a timing generation module, a memory module, and a conversion module. The control module receives an input control signal and generates a first control signal and a second control signal. The timing generation module is coupled to the control module and generates a timing signal according to the second control signal. The memory module is coupled to the control module and the timing generation module. The memory module receives and stores an audio signal and generates a memory signal according to the first control signal and the timing signal. The conversion module is coupled to the memory module and converts the memory signal to the output signal.

Abstract: A data processing apparatus is designed for simulating an acoustic characteristic of an acoustic space which contains a sound source for generating a sound and a sound receiving point for receiving the sound. In the apparatus, each of a plurality of characteristic control sections processes sound data and outputs the processed sound data. The characteristic control sections correspond to transmission paths which must exist in the acoustic space such that the sound generated from the sound source travels to the sound receiving point through the respective transmission paths. An instruction section provides a processing instruction of the sound data to each characteristic control section such that each characteristic control section processes the sound data according to the provided processing instruction to thereby execute the simulation of the sound traveling through the corresponding transmission path.

Abstract: A system used in a loudspeaker-room-microphone environment includes a microphone signal partitioner that divides a signal from a microphone into one or more divided portions. A reverberation energy estimator estimates reverberation energy in some of the divided portions of the microphone signal based on a loudspeaker signal. The estimated reverberation energy is processed to generate a dereverberated output signal.

Abstract: Scenes having at least one simulated sound source and simulated sound-reflecting objects are simulated by processing a direct-sound signal with at least one head-related transfer-function, thereby generating a simulated direct-sound signal, and generating simulated early-reflection signals from the simulated direct-sound signal, including simulating early reflections having incidence angles different from the incidence angle of the direct-sound signal. Externalization of the simulated sound source is enhanced.

Abstract: An audio signal processing system is configured to separate an audio signal into a dry signal component and one or more reverberant signal components. The dry signal component and the reverberant signal components can be separately modified and then recombined to form a processed audio signal. Alternatively, the dry signal component may be combined with an artificial reverberation component to form the processed audio signal. Modification of the reverberation signal component and generation of the artificial reverberation component may be performed in order to modify the acoustic characteristics of an acoustic space in which the audio signal is driving loudspeakers. The audio signal may be a pre-recorded audio signal or a live audio signal generated inside or outside the acoustic space.

Abstract: A method of identifying the user's own voice in a hearing instrument system and a hearing instrument system for performing such method is provided wherein a direct-to-reverberant ratio (DtoR) between the signal energy of a direct sound part (1a; 1b) and that of a reverberant sound part (2a, 3a; 2b, 3b) of at least a part of a recorded sound is used to assess wether the sound originates from the users own voice or not. This allows a very reliable detection of the users own voice in a hearing-instrument system. Further, a hearing-instrument system comprising an own-voice detector configured to perform such method is provided.

Abstract: The invention relates to a signal processing unit including an input for receiving an audio signal, and an early pattern generator connected to the input, for defining a predefined early pattern generation. The early pattern generator establishes an output having N directional components which are added to form a signal having N directional components. When representing each source output in a direction containing representation both directionality of the individual sound sources as well as the resulting directionality of the excited sound propagation may be contained and processed in a simple processing algorithm.

Abstract: The present invention provides an apparatus and a method for emulating audio. The apparatus comprises an echo device, a reverberation device, an audio source device and a first signal synthesizer. Wherein, the echo device receives an audio source signals and applies thereof for generating echo signal and feedback signal; the reverberation device receives the audio source signal and applies thereof in accordance with the feedback signal of the echo device for generating a reverberation signal; the audio source device receives the audio source signal and applies thereof for outputting a direct audio source signal; and the first signal synthesizer is used to receive and synthesize the echo signal, the reverberation signal and the direct audio source signal so as to output a first synthesized signal, which is an audio effect emulation signal.

Abstract: A Harmonic Generator for audio applications and a Pre-Amplifier circuit are combined to form a Harmonic Generator and Pre-Amplifier circuit. The Harmonic Generator is formed from a Buffer Circuit responsive to an input program signal for buffering the input program signal and for providing a buffered input program signal. A Modulator Circuit is coupled to receive the buffered input program signal and generates harmonics in response to changes in amplitude of the program input signal to provide a modulated input program signal. A Summing Circuit adds the buffered input program signal and the modulated input program signal to provide a composite modulated input program signal. The Pre-Amplifier is coupled to receive the composite modulated input programing signal and process the signal in three audio frequency bands to provide a compensated output signal that includes harmonics generated by the modulator.

Abstract: An device and method of generating environmental reverberation effects for digital audio signals is presented. The device includes a reverberation controller. The reverberation controller pre-processes one or more predetermined characteristics of a first audio signal to produce a pre-processed signal and generates a plurality of delayed outputs from the pre-processed signal, each output having a predetermined delay. The reverberation controller also produces a plurality of reflection outputs from the plurality of delayed outputs and combines the plurality of reflection outputs to produce a second audio signal having a desired reverberation response.

Abstract: The present invention is directed to creating sound reverberation. In accordance with a preferred embodiment, an apparatus estimates an impulse response for use in reproduction of a sound in a desired acoustic space. In particular, the apparatus collects space information concerning an acoustic space and point information indicating positions of a sound generation and reception points in the acoustic space, estimates a set of acoustic ray paths of the sound traveling from the sound generation point to the sound reception point, acquires directivity information indicating an acoustic directivity of the sound generation point and the sound reception point, estimates an acoustic intensity of each acoustic ray path and weights each acoustic intensity by the acquired directivity information, and determines the impulse response based on directions of the respective acoustic ray paths toward the sound reception point and the weighed acoustic intensities of the respective acoustic ray paths.

Abstract: A method and device for creating late reverberation in the frequency domain using spectral magnitude decay. The method applies a frequency domain representation of an input signal to a frequency domain accumulator having a delay, an adder and a frequency-dependent magnitude attenuation. The attenuated magnitude is combined with an artificial phase signal to produce one or more output signals. The method enables detailed control of late reverberation decay times as a function of frequency.

Abstract: A method and apparatus for extracting and changing the reverberant content of an input signal is provided. The method of estimating an impulse response for a reverberant system includes sampling a signal from a reverberant system, having an impulse response, into a sequence of blocks, for each block of the sequence, determining a magnitude in dependence upon a magnitude of a previous block, forming an estimate of the impulse response using determined magnitudes as parameters for a finite impulse response filter.

Abstract: A system including a buffer, a feedback loop configured to generate a fractional delay from a ratio of a first number of samples written into the buffer to a second number of samples read from the buffer, and a variable fractional delay filter configured to generate an output sample using a plurality of input samples and the fractional delay is provided.

Abstract: An apparatus and method of enabling equalization of a room parameter with a simple structure and low cost in an audio system using an acoustic transducer array. The apparatus to equalize a room parameter includes an audio signal generation unit to generate two identical audio signals, an audio signal modification unit to modify the audio signals, a sound beam generation unit to transform each of the two modified signals into a sound, to directly transmit one sound to the position of a listener, and to transmit the other sound to the position of the listener such that the other sound is reflected from a reflecting object and transmitted to the position of the listener, a sound magnitude determination unit to collect a sound generated through synthesis of the two sounds at the position of the listener, and a control unit to obtain a delay constant and a filter coefficient.

Abstract: A method and apparatus for estimating volume of an enclosed space (“room”) based on measured acoustic parameters. The method includes the steps of: measuring an acoustic impulse response of the enclosed space. From the acoustic impulse response, the method calculates the parameters: mean square pressure of reverberant sound; mean square pressure of direct sound; arrival time of direct sound; and a reverberation time parameter (T60). From those parameters, a volume estimate is calculated based on an acoustic diffuse field theoretical model.

Abstract: A TSP signal that is a measurement signal is generated N times by a signal generating portion. The generated TSP signals are supplied to a steel-echo apparatus. Reverberation generated by the steel-plate echo apparatus corresponding to the TSP signals is arranged N times corresponding to the start points of the TSP signals and synchronously added by a synchronously adding portion. Generating impluse signal is convoluted by an inverse function to TSP signal. The added result of the synchronously adding portion is supplied to an impulse response converting portion. By dividing the added result by the inverse characteristics of the TSP signals, the added reverberation is converted into an impulse response. The impulse response data obtained by the converting portion is recorded to for example a CD-ROM and supplied to a reverberator. The reverberator reproduces an impulse response recorded on the CD-ROM and performs a convolution calculation process for the input signal.

Abstract: An in-line early enhancement generation system comprises one or more microphones positioned close to one or more sound sources so as to detect predominantly direct sound, an early reflection generation stage which generates a number of delayed reproductions of the microphone signals and which has unitary power gain whereby the stability of the system is independent of the delay times and amplitudes, and a number of loudspeakers placed to broadcast said early reflected energy into the room.

Abstract: A reverberation apparatus is designed for generating a reverberation sound from an input sound with use of an impulse response based on an instruction. In the reverberation apparatus, a first storage section stores first impulse response data representative of a first impulse response. A second storage section stores second impulse response data representative of a second impulse response which is different from the first impulse response represented by the first impulse response data. A new data creating section operates based on the first impulse response data and the second impulse response data for creating new impulse response data representative of a new impulse response in accordance with the instruction.

Abstract: A reverberation processor includes a chain of absorbent all-pass filters and an absorbent delay line. The decay time can be precisely controlled by controlling the magnitude of the attenuation in the absorbent delay lines. Further, each absorbent delay line includes a low-pass filter for controlling the decay time at a particular high frequency.

Abstract: An acoustic modeling system and an acoustic modeling method use beam tracing techniques that accelerate computation of significant acoustic reverberation paths in a distributed virtual environment. The acoustic modeling system and method perform a priority-driven beam tracing to construct a beam tree data structure representing “early” reverberation paths between avatar locations by performing a best-first traversal of a cell adjacency graph that represents the virtual environment. To further accelerate reverberation path computations, the acoustic modeling system and method according to one embodiment perform a bi-directional beam tracing algorithm that combines sets of beams traced from pairs of avatar locations to efficiently find viable acoustic reverberation paths.

Abstract: The invention relates to a system for generating sounds with a left front loudspeaker which converts an electric signal of a left audio channel and with a right front loudspeaker which converts an electric signal of a right audio channel. Such a system for generating sounds in a room is known. The electric signals are processed such that spatial rear loudspeakers are dispensed with and the front loudspeakers generate a surround or spatial sound. According to the invention, a virtual filter generates virtual sound bodies laterally of a listener in a region between 80 and 100 degrees. Experiments have shown that the human ear is most sensitive to surrounding sound when sound sources are arranged laterally of a listener's head or when spatial sound is generated laterally of a listener.

Abstract: A process for enhancing ambience in an audio signal output that is derived from an audio signal input in a dual channel audio ambience extraction circuit. The process includes cross-coupling of audio signals in one channel with audio signals in another channel. Each of the cross-coupled signals is attenuated and delayed by an adjustable time period that is within a haas delay time and is then applied in the feedback path to a summing input of an opposite channel. At the summing input, the signals are mixed with subsequent audio signal inputs to that channel. All of the attenuated and delayed signals are continuously applied to outputs of the extraction circuit during the cross-coupling process. The output signals comprise the original signals plus delayed and attenuated reproductions of the original signal along with continuing signals that are submitted to the extraction circuit subsequent to the initial signals.

Abstract: An apparatus and method simulates more accurately the natural statistics of a physical reverberation process. A new filter design is provided having a comb shaped group delay. Gain minimums at a plurality of frequencies are combined with a delay line to create a constant reverberation time independent of frequency while allowing for temporal spreading. In addition, the connection topology between the plurality of energy transmission networks is temporally randomized to facilitate energy distribution within the reverberation apparatus. Both the temporal and spectral responses are actively changed on each iteration of the energy recirculation. By making the response have a high echo density and a lack of spectral coloration in the decay, the illusion of a natural process is enhanced.

Abstract: An audio system includes a circuit (12) for processing an audio signal, which circuit (12) having an input (20) for receiving the audio signal and an output (26) for supplying an output signal. The circuit (12) further includes a harmonics generator (22) coupled to the input (20) for generating harmonics of the audio signal and an adding circuit (24) coupled to the input (20) as well as to the harmonics generator (22) for supplying a sum of the audio signal and the generated harmonics to the output (26). The harmonics generator (22) includes an integrator (34) for integrating the audio signal and, coupled thereto, a resetting circuit (36) for resetting the integrator (34) at resetting times.

Abstract: A method for adjusting a time delay between a first audio signal and a second audio signal is disclosed. The method includes generating the first audio signal from a buffer as a first data stream and generating the second audio signal from the buffer as a second data stream after an initial time delay. The method further includes receiving the first data stream at a first sample rate converter at a first consumption rate and generating a first output data stream at an output sample rate. The second data stream is received at a second sample rate converter at a second consumption rate and the second sample rate converter generates a second output data stream at the output sample rate. One of the first and second consumption rates are changed so that the time delay between the first and second output data streams is adjusted over time from the initial time delay. A system for adjusting the time delay between a first audio signal and a second audio signal is also disclosed.

Abstract: A loudspeaker unit which requires no particular procedure for correction of the acoustic characteristic even if the installation environment of the loudspeaker unit changes, and which can correct, in addition to the frequency characteristic, a sound lag and a phase shift ascribable to the reverberation and an echo of a sound. The loudspeaker unit picks up a sound regenerating from the loud speaker with a microphone, and compares in real time a sound from a sound source with a regenerative sound, referring to a difference therebetween, with reference to the characteristic at an optional frequency and the characteristic of the reverberation or the echo each including the delay time, respectively, and corrects the signal to be sent to the loudspeaker by the result of arithmetic.

Abstract: A head-related impulse response to describe sound signals in a spatial environment is shown to accurately approximate three-dimensional sound data using limited computations, and can also be transformed for ease of computation. The head-related impulse response and disclosed computational methods thereof can be used to produce three-dimensional sound via a method described and refined. Implementations of the method can be used for applications with one or more sound sources, which may or may not have reflective information, and reproduced for a single listener or for multiple listeners.

Abstract: An improved reverberation processor includes a technique for changing environmental parameters without causing disturbing audio artifacts. The technique includes the steps of sequentially changing the read pointers of different delay lines. Additionally, for each delay line a level control variable is ramped down prior to changing the read pointer and then ramped back up. The reverberation processor also provides means for producing and controlling a repeating echo in the reverberation decay, as well as adjusting the diffusion (or echo density) of the reverberation.

Abstract: An audio processing apparatus that includes a first filter for converting n-channel (n?1, positive integer) audio signals input from at least one signal source into two-channel signals, a pair of second filters to which two-channel output signals from the first filter means are input and which have transfer functions that are not correlated, and an output unit for supplying a pair of output signals from the pair of second filters to left and right loudspeaker units of a headphone device.

Abstract: A method and apparatus for processing sound sources to simulate environmental effects includes source channel blocks for each source and single reverberation block. The source channel blocks include direct, early reflection, and late reverberation blocks for conditioning the source feeds to include delays, spectral changes, and attenuations depending on the position, orientation and directivity of the sound sources, the position and orientation of the listener, and the position and sound transmision and reflection properties of obstacles and walls in a modeled environment. The outputs of the source channel blocks are combined and provided to single reverberation block generating both the early reflections and the late reverberation for all sound sources.

Abstract: An in-plane position sensor detects positions along two directions or X-Y directions to produce a pair of position signals concurrently. The pair of position signals controls control parameters which influence upon a variety of effecters and addition processor which are implemented in a digital arithmetic unit (DSP).

Abstract: A wide frequency range signal from a test sound source 1 is reproduced successively by a plurality of speakers 101 to 104, and the reproduced sound is detected by a plurality of microphones 111 and 112, after which the frequency characteristics thereof are obtained at FFTs 4b and 4c, while obtaining the frequency characteristics of the wide frequency range signal at an FFT 4a. A high frequency range level is normalized with a low frequency range level, and a determination section 8 compares the normalized value with a reference value stored in a reference value storage section 9 to determine the number and positions of people in the sound field. At the same time, the transfer functions between the speakers and the microphones are calculated at transfer function calculators 10a and 10b, and impulse responses are obtained at IFFTs 12a and 12b, after which a reverberation time calculator 13 calculates the reverberation time based on the impulse responses.

Abstract: Disclosed is a method designed to stimulate the presence of one or more sound sources in virtual positions in three-dimensional space, in the perception of a listener occupying a real physical space, by means of suitable drive loudspeakers whose position is not perceived by the said listener, and a device for the implementation of the said method.

Abstract: A reverberation processor is provided comprising a plurality of delay lines implemented in delay line memory, and hardware operable to introduce an echo effect into the reverberation decay and to provide a control parameter to control salience of the echo effect. The delay lines are operable to generate a reverberation decay in response to an input signal. In one embodiment, an increase and a decrease in the salience of the echo effect is dependent upon the control parameter. The control parameter may provide continuous control over the salience of the echo effect and the echo effect may be embedded within the reverberation decay. In one embodiment, a range of delay lengths across which the plurality of delay lines is distributed is reduced to produce a repeating echo effect. The invention extends to method of and software product for providing reverberation.

Abstract: Included in the embodiments of the present invention is a technique to detect sound with a sensor to generate a corresponding sound signal and iteratively determine two or more values with a maximum likelihood function for evaluation of reverberation time. One of these values corresponds to a time constant parameter, and another of these values corresponds to a diffusive power parameter. An estimate representative of the reverberation time is further provided as a function of an order-statistics filter.

Abstract: In a reverberation apparatus, a storage section stores a directional characteristic representing a directivity of generated sound at a sound generating point. A position determining section determines a position of the sound generating point within an acoustic space on the basis of an instruction from the user. An orientation determining section determines an orientation of the sound generating point based on the determined position thereof. An impulse response determining section determines an impulse response for each of sound ray paths along which the sound emitted from the sound generating point travels to reach a sound receiving point, in accordance with the directional characteristic of the generated sound and the orientation of the sound generating point. A calculation section performs a convolution operation between the impulse response and an input audio signal so as to apply thereto the acoustic effect.

Abstract: An audio system includes a circuit (12) for processing an audio signal, the circuit (12) having an input (20) for receiving the audio signal and an output (26) for supplying an output signal. The circuit (12) further includes a harmonics generator (22) coupled to the input (20) for generating harmonics of the audio signal and an adding circuit (24) coupled to the input (20) as well as to the harmonics generator (22) for supplying a sum of the audio signal and the generated harmonics to the output (26). The harmonics generator (22) includes an integrator (34) for integrating the audio signal, and, coupled thereto, a resetting circuit (36) for resetting the integrator (34) at resetting times.

Abstract: N correlated signals are processed by N pre-filters whose transfer characteristics have different zero points, then the processed signals are input into an N-input M-output linear FIR system, and its transfer characteristics are estimated from its response outputs and the processed signals from the pre-filters.

Abstract: A reflection sound generator has a first filter of Finite Impulse Response (FIR) type that is provided with a first set of parameters representing a first distribution pattern of reflection sounds, and a second FIR-type filter provided with a second set of parameters representing a second distribution pattern of additional reflection sounds. The first distribution pattern has a time length sufficient to cover an initial reflection sound and subsequent reverberant reflection sounds which are distributed at intervals along the time. The first filter executes convolution operation of sample data of an input sound by the first set of parameters to generate first data containing a sequence of reflection sounds of the input sound. The second filter executes convolution operation of the first data by the second set of parameters to generate second data containing additional reflection sounds which fill the intervals of the reflection sounds of the input sound.

Abstract: A sound-field production apparatus includes a direct-sound-generating unit for generating a direct sound from a sound signal supplied to the direct-sound-generating unit and a reflected-sound-generating unit for generating a sound serving as a substitute for a sound obtained as a result of reflection of the direct sound at a level lower than the direct sound with a timing lagging behind the direct sound from the sound signal supplied to the reflected-sound generating unit. The sound-field production apparatus makes it possible to reproduce an impressive sound field that gives a better feeling of presence on the scene as a sound field in a movie theater or the like does by creating a sound-field space having an effect of sound-image broadening.

Abstract: A music effector apparatus is constructed for processing a digital audio signal to produce a music sound while imparting acoustic effects to the music sound by executing microprograms corresponding to the acoustic effects. In the music effector apparatus, a digital signal processor has a process capacity only sufficient to execute a limited number of microprograms in parallel to each other to create corresponding acoustic effects in production of the music sound. An operating panel is manually operable when changing the acoustic effects during the production of the music sound for setting an acoustic effect to a zero level so that the corresponding microprogram is made inactive, and for setting another acoustic effect to other level than the zero level so that the corresponding microprogram is made active.

Abstract: A system and method for acoustic modeling partitions an input 3D spatial model into convex cells, and constructs a cell adjacency data structure representing the neighbor relationships between adjacent cells. For each sound source located in the spatial environment, convex pyramidal beams are traced through the input spatial model via recursive depth-first traversal of the cell-adjacency graph. During beam tracing, a beam tree data structure is constructed to encode propagation paths, which may include specular reflection, transmission, diffuse reflection, and diffraction events, from the source location to regions of the input spatial model. The beam tree data structure is then accessed for real-time computation and auralization of propagation paths to an arbitrary receiver location.

Abstract: An audio apparatus has a main speaker SPA receiving an audio signal from a sound source 10 and outputting sound of the sound source; sound effect audio signal processing circuits 12B, 12C, etc. for performing signal processing on the audio signal from the sound source 10 to generate a sound effect audio signal for reproduction of a sound effect required of the sound of the sound source; and sound effect speakers SPB, SPC, etc. each receiving the sound effect audio signal and outputting the sound effect required of the sound of the sound source.

Abstract: A signal processor with a small circuit size, a simplified arrangement, and reduced cost and which can generate the same sound field as that in a case of listening to a reproduced sound using speakers to a location outside of a head of the listener has FIR filters and reverberation adding circuits. At an addition point provided to a tap midpoint of one FIR filter, a delay output signal of a tap end from another FIR filter is multiplied by a predetermined multiplication coefficient and then added. Output of one FIR filter is subjected to reverberation adding processing by the reverberation adding circuits and then output of yet another FIR filter is added to the above output to obtain an output signal. Therefore, even if the FIR filters with short tap lengths are used, long impulse response can be reproduced and a scale of signal processing can be reduced substantially.

Abstract: In a signal generation circuit for a compound acoustic actuator that generates a sound and a vibration in response to a frequency of a signal input to the compound acoustic actuator, the signal generation circuit comprising a plurality of signal data stored in a memory to generate a plurality of signals having mutually different frequencies, the plurality of signals at least including a signal, a frequency of which is equal to a resonant frequency causing the compound acoustic actuator to generate the vibration, a synthesizing means to synthesize a plurality of drive signals in accordance with the plurality of signal data so as to cause the compound acoustic actuator to generate the vibration, and a sweeping means to sweep the plurality of drive signals, repeatedly.

Abstract: A multi-channel reverberation system has a substantially constant multi-channel power gain at audio frequencies and comprises multiple signal inputs, one for each input channel, a number of comb filter networks connected to each signal input, each comb filter network including a feed forward stage, a cross-coupling network cross-coupling the comb filters to increase the reverberation echo density, and multiple signal outputs, one for each output channel. Preferably the feed forward stage of each comb filter provides a substantially constant multi-channel power gain at audio frequencies and the cross-coupling matrix is an orthogonal cross-coupling matrix cross-coupling a number of single channel allpass comb filters, positioned immediately before or after the delay lines, to create a multi-channel allpass comb filter with a unitary transfer function matrix at all frequencies.