Abstract: Embodiments pertain to a method for use with a musical piece, or part thereof, comprising: presenting, by at least one client device to a plurality of users, a corresponding plurality of sequences of musical symbols to be cooperatively played by the plurality of users using a plurality of instruments; capturing instrument outputs produced by the plurality of instruments to generate audio data descriptive of the captured instrument outputs; analyzing the audio data for determining a level of correspondence between the captured instrument outputs and the plurality of musical symbols presented to the plurality of users; and presenting, based on the level of correspondence, an adapted or non-adapted sequence of music symbols to at least one of the plurality of players.

Abstract: Vehicle (1), comprising a vehicle cabin (2) defining an acoustic space having specific aurally perceivable properties in a reference state, comprising an apparatus (3) for outputting an audio signal (4) in the vehicle cabin (2), characterized in that the apparatus (3) for outputting an audio signal (4) in the vehicle cabin (2) comprises an audio outputting device (6) configured to output an audio signal (4) in the vehicle cabin (2); a acoustic impression modifying device (8) configured to generate an acoustic space modification signal (9) allowing for modifying the aurally perceivable properties of the acoustic space defined by the vehicle cabin (2) with regard to a reference state.

Abstract: An example method of operation may include detecting an acoustic stimulus via active beams associated with at least one microphone disposed in a defined space, detecting loudspeaker characteristic information of at least one loudspeaker providing the acoustic stimulus, transmitting acoustic stimulus information based on the acoustic stimulus to a central controller, and modifying, via a central controller, at least one control function associated with the at least one microphone and the at least one loudspeaker to minimize acoustic feedback produced by the loudspeaker.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media for generating echo recordings. The system receives, by an autoencoder, an audio signal representation that represents an audio signal and a target echo embedding that comprises information about a target room. The autoencoder comprises an encoder and a decoder. The system generates, by the encoder, a content embedding and an estimated echo embedding. The system generates, by the decoder, an echo recording representation based on the content embedding and the target echo embedding.

Abstract: A system for calculating reverberation, comprising a computer that receives a signal input; stores an impulse response signal; reads out the impulse response signal corresponding to the input signal; converts an input signal and an impulse response signal into an input signal in the frequency domain and an impulse response signal in the frequency domain; samples the input signal and in the frequency domain and the impulse response signal in the frequency domain to an input signal and an impulse response signal after sampling; sets one of values of signals included in the impulse response signal after sampling to zero to obtain an impulse response signal after zero-value processing; performs a convolution operation on the impulse response signal after zero-value processing to obtain a reverberant signal in the frequency domain; and converts the reverberant signal in the frequency domain into a reverberant signal in a time domain.

Abstract: According to one aspect, there is an electronic musical instrument including operators which includes a first operator corresponding to a first pitch and a second operator corresponding to a second pitch, and a sound source configured to generate a first output signal from a first close loop and a second output signal from a second close loop, to generate an integration signal, and return a subtraction signal, the subtraction signal being obtained by subtracting a signal circulating through the first close loop from the integration signal, to the first close loop, and then output a musical sound signal including signal components corresponding to the second pitch as a musical sound signal corresponding to the first pitch.

Abstract: Provided are a method and device for processing a music file, a terminal and a storage medium. The method comprises: in response to a received sound effect adjustment instruction, acquiring a music file, the adjustment of which is indicated by the sound effect adjustment instruction; carrying out vocals and accompaniment separation on the music file to obtain vocal data and accompaniment data in the music file; carrying out first sound effect processing on the vocal data to obtain target vocal data, and carrying out second sound effect processing on the accompaniment data to obtain target accompaniment data; and synthesizing the target vocal data and the target accompaniment data to obtain a target music file.

Abstract: A sound signal processing method includes: obtaining a plurality of sound signals respectively collected by a plurality of microphones arranged in a space; adjusting respective levels of the plurality of sound signals in accordance with respective positions of the plurality of microphones; mixing the plurality of sound signals having the adjusted respective levels to thereby obtain a mixed signal; and generating a reflected sound by using the obtained mixed signal.

Abstract: An example method may include recording, via a microphone of a first playback device, audio played by a second playback device in accordance with a calibration setting. The method may further include, based on the recorded audio, determining that the calibration setting is invalid. The method may further include, in response to determining that the calibration setting is invalid, sending an indication that the calibration setting is invalid. This disclosure also includes example non-transitory computer readable media and playback devices that are related to the example method.

Abstract: Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal.

Abstract: A signal processing device is designed to automatically calculate a transfer characteristic representing sound-box resonance of a guitar due to acoustic excitation of vibration which may occur due to white noise. White noise emitted toward the guitar causes vibration propagating via strings so as to produce an audio signal via a pickup. A transfer characteristic is calculated based on an audio signal and a white-noise signal. A filter performs convolution, using the transfer characteristic, on audio data representing user's playing sound of the guitar, thus reproducing sound-box resonance indicating distinctive peaks which may appear in a low-frequency range of guitar's sound. It is possible to store a plurality of transfer characteristics in memory, whereby any user may be allowed to select a desired transfer characteristic among transfer functions stored in memory or to utilize a transfer characteristic actually calculated by the signal processing device.

Abstract: A sound capture device is affixed to an acoustic instrument to capture the natural sound output of the instrument. The captured sound signal is routed to an electronic sound augmentation system that is configured to augment the captured sound with spatial sound effects such as reverb, echo, delay, etc. The processed and augmented sound is then reproduced via a vibrating driver that has been affixed to the body of the acoustic instrument. This creates a situation where the body of the musical instrument, responding to a series of vibrations produced by the vibrating driver, acts as a speaker component, reproducing a rich augmented sound output that comprises the sum of the sound produced by the original sound production capabilities of the acoustical instrument plus the added augmented or enhanced sound effects.

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: A method and apparatus of generating a sound field effect is provided. The sound field effect generating apparatus may generate a frequency coefficient that is frequency-transformed from a direct signal, may generate a reflection signal from the frequency coefficient, may generate an output signal using the frequency coefficient and the reflection signal, and may perform an inverse-frequency transform of the output signal.

Abstract: A piano with an electronic tone generator capable of outputting musical sound signals of a reverberation and/or a resonance, in accordance with whether an external output is performed or not.

Abstract: The sound field reproducing apparatus includes: a signal obtaining unit which obtains input audio signals of plural channels; an effect adding unit which gives an effect on an output to a sound field; and an effect amount control unit which controls an amount of the effect based on a characteristic of the input audio signal. The sound field reproducing apparatus receives input audio signals of plural channels from a medium such as a DVD, and gives a sound effect to them to output. The amount of the effect given to the input audio signal is controlled based on the characteristic of the input audio signal, especially the cross correlation of the reverberation component between the channels.

Abstract: A surround experience for headphones can be created using various techniques and tools. The described techniques and tools can be applied separately or in combination. For example, a surround experience can be created by splitting input audio channel signals into high-pass and low-pass signals. The low-pass signals can undergo cross-mixing and artificial reverberation. The artificial reverberation can apply different delay profiles to the low-pass audio signals. The high-pass and low-pass audio signals can be combined to generate output audio channel signals. A surround experience can be crated using a system comprising an input module, a crossover module, a channel cross-mixing module, a reverberation module, and an adder module.

Abstract: An electronic reverberation system employs a processor to produce a plurality of delays samples that are added to a direct signal to produce reverberant sound. The disclosed system generates or employs a list of gain value pairs that are produced based on control settings or are provided as fixed coefficients. The processor generates reverberation samples by applying these coefficients to delay samples and summing their amplitudes to produce reverberation waveform samples. The reverberation waveform samples are added to the direct signal.

Abstract: A product-sum operation circuit has delay circuits of the first to the (n?1)th stage for delaying musical tone data, multiplying circuits 60-6(n?1) for multiplying the musical signal data or the delayed musical signal data output from the delay circuits by impulse response coefficients, and adders 71-7(n?1) for summing up data output from the multiplying circuits. The product-sum operation circuit is provided with a feed back circuit. The feed back circuit includes a multiplying circuit 80 that receives the delayed data from the delay circuit at the (n?1)th stage and multiplies the received data by a multiplication coefficient, and an adder 81 for adding data from the multiplying circuit 80 to the delayed data from the delay circuit at the “p”th stage.

Abstract: An artificial-reverberation generating device comprises in one embodiment a convolution engine in association with an impulse-response synthesizer. The impulse-response synthesizer preferably comprises a noise synthesizer and a control means for controlling one or more parameters of the noise synthesizer, the parameters corresponding to adjustable characteristics of the generated reverberation. The noise synthesizer is preferably a pseudo-random number generator, which is multiplied by the output signal of a density generator. The density generator output signal advantageously takes the form of a series of spikes of variable duration and time-interval spacing. The multiplier outputs feed a phase-correlation stage, followed by a time-variant filter stage and a time-variant amplifier stage. The output of the time-variant amplifier stage forms the output of the impulse-response synthesizer and is used to feed impulse-response information to the convolution engine.

Abstract: A reverberation effect adding device comprising a first convolution circuit which in turn comprises FIR filters (80-1 to 80-4) and an adder (accumulator) (81) which adds outputs from the FIR filters, a moving average circuit (82) which receives musical sound waveform data delayed by a predetermined number of stages in the first convolution circuit and which outputs averaged second musical sound waveform data obtained by sampling at a second sampling frequency lower than the first sampling frequency, a second convolution circuit which in turn comprises FIR filters (80-5 to 80-28) which sequentially receive the second musical sound waveform data obtained by sampling at the second sampling frequency and an adder (accumulator) (83), an interpolator (84) which receives an output from the adder (83) of the second convolution circuit, calculates an interpolated value of the output value from the adder (83), and provides the output from the adder (83) and the interpolated value from the interpolator (84) sequentially

Abstract: A dry/wet bit for controlling dry and wet components of an output sound during processing of an input sound is provided. The bit is configurable by a program to indicate when to reverse the dry and wet components of the output sound. When the bit has a first value, the dry component is calculated by modifying the input sound by an attenuation factor, and the wet component is calculated by providing to a reverberation filter as input the dry component modified by a reverberation factor. When the bit has a second value, the wet component is calculated by providing to the reverberation filter as input the input sound modified by the attenuation factor, and the dry component is calculated by modifying the input to the reverberation filter by the reverberation factor.

Abstract: A digital piano apparatus includes an input unit including a plurality of keys and a plurality of pedals, a sound source unit, a sound field synthesizing unit, and a speaker unit. The sound field synthesizing unit includes virtual sound source spot signal generating means for generating a plurality of virtual sound source spot signals of a number equivalent to the number of output channels in the speaker unit, the plurality of virtual sound source spot signals being used for forming reproduced sound fields of a grand piano on the basis of a plurality of tone signals supplied from the sound source unit; mixing means for mixing each virtual sound source spot signal with the remaining virtual sound source spot signals; and sound effect means for applying a sound effect by signal processing using impulse responses to the virtual sound source spot signals.

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 transmission 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 electronic circuit for adding the effects of reverberation to a high impedance signal produced from an external audio source such as a guitar pickup or a high impedance microphone.

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: 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: 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 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: An effect system stored a specified segment of an audio signal and repeats the segment during a specified interval. The segment length may be defined in advance as a musical time interval and the interval in which it is repeated may be defined in advance as a musical time interval or as a number of repetitions. The beginning of the segment may be defined in advance as a location in musical time within the audio signal, or may be indicated manually during performance.

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 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 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: Reverberation characteristics of an original sound field and a reproduction sound field are measured and a difference between the reverberation characteristics is obtained. A reflected sound pattern added to reproduction sound is calculated so that the reverberation characteristics of the reproduction sound field approximate to the reverberation characteristics of the original sound field from the difference between the reverberation characteristics and is added and reproduced in the reproduction sound field. A frequency is divided into predetermined bands and a reflected sound pattern to be added is obtained similarly in the respective bands and is added to the corresponding frequency band of the reproduction sound and reproduced. Measurement of the reverberation characteristics is obtained by inputting an impulse signal to a speaker and collecting a generated sound wave by a microphone provided in a position corresponding to the head of a listener and analyzing the sound wave by an analyzer.

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: 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 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: A reverberation generation processor. The reverberation generation processor generates a specific auditory signal according to a predetermined virtual environment having characteristics of spatial dimension, sound reflection, and decline. The natural sequence generator generates a natural sequence having a predetermined number of items according to the characteristics of the virtual environment. The sequence adjusting device adjusts gains of the items and a time scale between the items of the natural sequence and outputs an audio signal. The filter processor filters a predetermined frequency band of the audio signal according to the characteristics of the virtual environment and divides the filtered audio signal into a high frequency audio signal and a low frequency audio signal. The high frequency reverberation generator transforms the high frequency audio signal to a high frequency reverberation signal.

Abstract: A method of producing reverberation effects is disclosed. A filter is implemented for modeling early acoustic reflections in response to an input signal using a first processor, the filter includes a delay buffer of a selected length and having a selected number of taps for tapping samples of corresponding amounts of delay and a summer for summing the tapped samples to generate a filter output signal. A reverberator is implemented for modeling late acoustic reflections using a second processor, the reverberator receiving the filter output and generating a plurality of output signals.

Abstract: An electronic musical instrument according to the invention stores a difference tone identifying table which associates musical sounds of different pitches with a difference tone perceived from the musical sounds of different pitches. Receiving an instruction to simultaneously produce musical sounds of different pitches, the electronic musical instrument extracts a difference tone corresponding to the musical sounds of different pitches from the difference tone identifying table and generates then outputs a sound signal corresponding to the extracted difference tone and the musical sounds of different pitches.

Abstract: An apparatus estimates an impulse response for use in reproduction of a sound in a desired acoustic space. In the apparatus, a first acquisition section acquires space information indicating a spatial shape of the acoustic space and an acoustic reflectivity of a boundary surface enclosing the acoustic space. A second acquisition section acquires point information indicating positions of a sound generation point and a sound reception point set in the acoustic space. An estimation section estimates a set of acoustic ray paths of the sound traveling from the sound generation point to the sound reception point based on the acquired space information and the point information. A third acquisition section acquires directivity information indicating an acoustic directivity of the sound generation point and the sound reception point. A weighting section estimates an acoustic intensity of each acoustic ray path, and weights each acoustic intensity by the acquired directivity information.

Abstract: Reverberation characteristics of an original sound field and a reproduction sound field are measured and a difference between the reverberation characteristics is obtained. A reflected sound pattern added to reproduction sound is calculated so that the reverberation characteristics of the reproduction sound field approximate to the reverberation characteristics of the original sound field from the difference between the reverberation characteristics and is added and reproduced in the reproduction sound field. A frequency is divided into predetermined bands and a reflected sound pattern to be added is obtained similarly in the respective bands and is added to the corresponding frequency band of the reproduction sound and reproduced. Measurement of the reverberation characteristics is obtained by inputting an impulse signal to a speaker and collecting a generated sound wave by a microphone provided in a position corresponding to the head of a listener and analyzing the sound wave by an analyzer.

Abstract: A incoming-call tone generation device comprises: a first storage unit 1, for storing an effect calculation formula for incoming-call tone data; a second storage unit, for storing incoming-call tone data that consist of data for original sounds; a controller, for reading incoming-call tone data from the second storage unit and transmitting the data to a digital signal processor 4, and for controlling a D/A converter 5; the digital signal processor 4, for calculating an effect pattern and a sine wave using the calculation formula read from the first storage unit, and for performing an envelope process for the incoming-call tone data; the D/A converter 5, for converting, into analog data, digital incoming-call tone data obtained during the envelope process, and for outputting an analog incoming-call tone; and a loudspeaker 6 through which an analog incoming-call tone is released.

Abstract: Reverberation imparting functions as are obtained by a mechanical type reverberation apparatus employing coil spring delaying elements are simulated by an electric circuit configuration. Three delay loop circuits are individually provided, each including a first and a second delay circuit and delaying and circulating an input signal to thereby simulate the propagation of vibrations through each individual coil spring. Multiplier circuits are provided to individually supplying the out puts from the first and the second delay circuit in one delay loop circuit to the input sides of the second and the first delay circuit in two other delay loop circuits after controlling the signal characteristic to thereby simulate the propagation of vibrations through the supporting plates for the coil springs.

Abstract: A sound generating system produces tones from an audio signal as if tones originally produced from an acoustic guitar are converted to an electric signal through a microphone; the electric signal is produced from vibrations of the strings through a pickup unit, and is converted to digital signals; a digital signal processor simulates reverberation generated through a resonator of the guitar and direct tones directly produced from the strings through plural sorts of digital data processing; the signal component representative of the reverberation is added to the signal component representative of the direct tones for producing the audio signal; the digital signal processor simulates the direct tones through the function of an all-pass filter so that the reproduced sound has the timbre approximate to the timbre of the original tones.

Abstract: A method and apparatus for generating a simulated reverberation characteristic in an input audio signal having a plurality of channels, each channel transmitting time-synchronized data elements. The method and apparatus imparts a unique delay to each element of a set of time-synchronized data elements and thereafter delays the data element set through a series of delay elements. Each of a plurality of early reflection taps reads in and outputs a unique linear combination of the delayed data elements. The delayed elements are selectively input to at least one summing filter incorporating a plurality of comb filters. The output of the at least one summing filter is selectively combined based upon a summing scheme to provide a plurality of sum output elements. The sum output elements are combined with the early reflection tap output and a preselected combination of the time-synchronized data elements of the input signal to create a simulated reverberation signal.

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.