Abstract: A sound or music synthesizer includes a reverberation simulator having a substantially reduced volatile storage, random access memory, or buffer size in comparison to conventional reverberation simulators by decimating the sound signal prior to applying the sound signal to a reverberator and then interpolating the sound signal generated by the reverberator to restore the sample frequency. The substantial reduction in buffer size enables the usage of the reverberator in low-cost, reduced size and single-chip environments.

Abstract: A system and method is provided for performing sound synthesis with delay-based special effects which may be algorithmically implemented using one or more time-delay elements The system implements the time-delay elements by using system memory to store time-delay data. The system and method described herein utilizes the benefits of a high bandwidth I/O bus while mitigating the disadvantages introduced by having to arbitrate for a shared system bus. By using system memory for storing time-delay data, a more cost effective PC audio system can be produced.

Abstract: The manipulation system and apparatus receives electronic signals which are to be processed and recorded for playback as enhanced stereophonic audio signals from two laterally spaced loudspeakers in front of the listener. The system and apparatus processes those signals to produce a conditioning signal, such as which would be produced by virtual room boundaries, which is heard together with the original signals so that an enlarged listening area is perceived by the listener. By amplitude and phase control of the signal to the two real speakers, the system and apparatus provide a means for control over the enhanced sound field. This enhanced sound field is perceived by the listener as being contained within boundaries larger than those normally reproduced by stereophonic speakers.

Abstract: A method of hiding information in a host audio signal introduces one or more echoes into the signal. The separation in time between the host signal and an echo is associated with the value of a datum embedded in the signal. The identity of the embedded datum is determined by observing the delay between the host signal and the echo.

Abstract: In a reverberation effect imparting apparatus, sampled data of an input signal is written into a RAM 23 and the data is read out after an elapse of a predetermined time period, thereby imparting a reverberation effect. A storage area of the RAM 23 is divided into an area 1 for storing waveform data and an area 2 for imparting a reverberation effect. During evaluation, the waveform data is sequentially read out from the area 1 by using an absolute address, and sampled data of the waveform data is written into and read out from the area 2 by using a relative address, thereby imparting a reverberation effect.

Abstract: A karaoke apparatus is constructed to perform a karaoke accompaniment part and a karaoke harmony part for accompanying a live vocal part. A pickup device collects a singing voice of the live vocal part. A detector device analyzes the collected singing voice to detect a musical register thereof at which the live vocal part is actually performed. A harmony generator device generates a harmony voice of the karaoke harmony part according to the detected musical register so that the karaoke harmony part is made consonant with the live vocal part. A tone generator device generates an instrumental tone of the karaoke accompaniment part in parallel to the karaoke harmony part.

Abstract: An improved method for shifting the pitches of a tone is disclosed. It comprises: (a) subjecting a digitized original waveform to a whitening process using an all-zero filter (AZF) to obtain a whitened waveform; (b) resampling the whitened waveform at a desired scaling ratio to obtain a scaled and whitened waveform; (c) subjecting the scaled and whitened waveform to a coloring process using an all-pole filter (APF) to obtain a synthesized waveform. In a preferred embodiment, the all-zero filter performs the transformation function of: ##EQU1## and the all-pole filter performs the transformation function of: ##EQU2## wherein the a.sub.i 's and b.sub.i 's are linear predictive coefficients. The whitened waveforms can be compressed and stored as wavetables, which can be subsequently retrieved and decompressed before resampling.

Abstract: A DSP programming apparatus capable not only of reducing work load for software engineers at the time of doing DSP programming but also of modularizing commonly usable algorithms, as well as to provide a DSP device capable of storing programs prepared by the DSP programming apparatus. A module storage unit stores a plurality of modules on a function basis in advance, and modules are linked by combining on a display screen through a GUI (graphical user interface) function by an input section. The linked DSP programs are converted into a DSP microprogram in executable form by a microprogram preparing section, and the microprogram is then transferred to a DSP through a transfer section.

Abstract: A wideband assisted reverberation system has multiple microphones (M1-M3) to pick up reverberant sound in a room, multiple loudspeakers (L1-L3) to broadcast sound into the room, and a reverberation matrix connecting a similar bandwidth signal from the microphones (m) through reverberators to the loudspeakers (L). Preferably the reverberation matrix connects each microphone (m) through one or more reverberators to at least two loudspeakers (L) with cross-linking so that each loudspeaker (L) receives a signal comprising a sum of at least two reverberated microphone signals. Most preferably there is full cross-linking so that every microphone (m) through reverberators to every loudspeaker (L), so that each loudspeaker (L) receives a signal comprising a sum of reverberated microphone signals from every microphone (m).

Abstract: The present invention is concerned with an audio equipment capable of processing video signals and a method of displaying operation of the acoustic equipment. An input selector switch 1 is supplied with reproduced audio signals from devices connected to a tuner unit 2 and input terminals 3, and supplies a signal to a sound field processing DSP unit 4. A signal processed by the DSP unit 4 is applied through a muting switch 5 to an output terminal 6. An input selector switch 7 is supplied with reproduced video signals from devices connected to input terminals 8, and supplies a signal to a graphic controller (GDC) 9 which generates a video signal to display an image. One of a signal generated by the GDC 9 and an original video signal are selected by a switch 10, and applied to an output terminal 11. The GDC 9 also generates an image simulating a control panel. When a key switch 13 is operated, the display corresponding to the operated key in the simulating image is varied.

Abstract: A reverberation generating system stores parameter data representative of a series of timings and a series of sound intensities for an initial part of reverberation, and calculates a series of parameter data for the later part of reverberation by using a return map of the proportional constant for time intervals of the initial part inversely proportional to the square of lapse of time so that the reverberation generating system stores the parameter data for the initial part only.

Abstract: A method for the simulation of the acoustical quality produced by a virtual sound source and for the localizing of this source with respect to one or more listeners, and one or more original sound sources. This method consists in: 1) fixing values of perceptual parameters defining the acoustical quality to be simulated and values of parameters defining the localization of a virtual source, 2) converting these values into a pulse response described by its energy distribution as a function of the time and the frequency, 3) carrying out a context compensation so as to take account of an existing room effect, 4) obtaining an artificial reverberation from elementary signals so as to achieve a virtual acoustic environment in real time and control the localizing of the virtual source. This method can be used to modify sound signals coming from a real source, or to create sound effects on recording media.

Abstract: In order to change an electronic sound between a piano sound generated without manipulation of a damper pedal and a piano sound generated under manipulation of the damper pedal, an electronic keyboard musical instrument assigns two tone generating channels to a depressed key for generating a fundamental tone signal representative of the piano sound generated by striking the set of strings and a resonating tone signal representative of a resonating sound generated through the resonance, and the resonating sound signal is mixed with the fundamental tone signal under manipulation of a pedal corresponding to the damper pedal, thereby making the electronic sound close to the piano sound.

Abstract: Spatialization of soundfields is accomplished by filtering audio signals using filters having unvarying frequency response characteristics and amplifying signals using amplifier gains adapted in response to signals representing sound source location and/or listener position. The filters are derived using a singular value decomposition process which finds the best set of component impulse responses to approximate a given target set of impulse responses corresponding to head related transfer functions. Efficient implementations for rendering reflection effects, air absorption losses and other ambient effects, and for spatializing multiple sound sources and/or generating multiple output signals are disclosed.

Abstract: The apparatus of the invention calculates filter coefficients for controlling sound field and sound image, based on a plurality of first impulse response signals and a pair of second impulse response signals. The plurality of first impulse response signals indicate impulse responses from loudspeakers reproducing audio signals to both ears of a listener. The pair of second impulse response signals indicate impulse responses from a reference loudspeaker at a position at which a sound image is localized to both ears of the listener.

Abstract: In the method and apparatus for obtaining acoustic characteristics of sound of a broad frequency range of from, for example, 0 to 20 KHz in a relatively short time with high accuracy even though an inexpensive computer is used; multiple sound ray vectors are defined; a virtual space is defined by a polygonal boundary; the propagation history data of the vector is calculated and stored, the vector being reflected at the boundary; and, based on the data, as for each of the vectors, a transient response thereof at an observation point is added to a time-series numerical array and stored, the response being determined on the basis of the reflected vector and a velocity potential determined at the observation point by a micro-area element of the vector defined on the boundary.

Abstract: A reverberation network for processing an audio signal including a plurality of wave-ladder filters, each of which simulates a corresponding reflected audio signal from wave propagation in a selected direction, wherein each of the plurality of wave-ladder filters receives as input a signal derived from the audio signal; and an adder which combines the corresponding reflected signals of the plurality of wave-ladder filters to produce a reverberation signal.

Abstract: An acoustic image localization apparatus localizes an acoustic image of a plurality of tone color species having various timbres to a source point with respect to a listening point within a sound field. Desired ones of separate source points are specified, which are provisionally set throughout the sound field. Each of the tone color species is assigned to one or more of the specified source points such that each specified source point is assigned with a group of desired tone color species. Acoustic images of the respective groups are localized to corresponding ones of the specified source points. Further, one of separate listening points is designated, which is provisionally set relatively to the source points so that the acoustic images are localized with respect to the designated listening point.

Abstract: A karaoke system of an online type is comprised of a karaoke station for sounding a karaoke performance according to a performance data, and a host station for supplying the performance data to the karaoke station in response to a request. The host station has a transmitter unit for time-sequentially dividing the performance data into a plurality of parts, and for successively transmitting each part arranged according to progression of the karaoke performance in response to the request. The karaoke station has a receiver unit operative when a preceding one of the transmitted parts is received for commencing the karaoke performance based on the received preceding part while admitting a succeeding one of the transmitted parts in parallel to sounding of the karaoke performance to thereby maintain continuous progression of the sounding of the karaoke performance.

Abstract: A surround audio signal reproducing apparatus for reproducing a multi-channel audio signal to generate a surround sound effect is provided. The apparatus contains a surround decoder, a sub-woofer mixing section, a first rear amplifier, and a speaker system having a sub-woofer and a first rear speaker. The surround decoder inputs and decodes the multi-channel audio signal to produce a first rear signal and a sub-woofer signal. The sub-woofer mixing section inputs the sub-woofer signal and the first rear signal and produces a mixed first rear signal. The first rear amplifier inputs and amplifies the mixed first rear signal to produce an amplified first rear signal. Then, the speaker system extracts the sub-woofer signal from the amplified first rear signal and reproduces the sub-woofer signal via the sub-woofer. Furthermore, the speaker system extracts the first rear signal from the amplified first rear signal and reproduces the first rear signal via the first rear speaker.

Abstract: A telephone instrument creates spatially simulated sound signals from signals received from a telephone line. The monaural signals are separated to two left and right channels. In each channel, the signals are processed via a direct path, an early reflection path including a finite impulse response filter and a reverberant decay path including all-pass filter. In each channel, the outputs from the three paths are summed with different weights.

Abstract: By using special frequency response manipulation in the difference channel of a stereo signal, the stereo image will appear to extend beyond the actual placement of the loudspeakers. This is accomplished by shaping the difference channel response to simulate the response one would be subjected to if the sources were physically moved to the virtual positions. The circuit includes a summing and high frequency equalization circuit to which the left and right stereo signals are applied, and a difference forming and human ear equalization circuit also to which the left and right stereo signals are applied. The outputs from these circuits are cross-coupled to form left and right channel outputs.

Abstract: A wavetable audio synthesizer with means for eliminating zipper noise caused by large volume increments, especially at slower rates of volume increment. The wavetable audio synthesizer includes shift circuitry which stores the value of volume increment in binary format and shifts the value right when the shift circuitry is enabled. Shifting the increment value right divides the value by an amount based on the number of bit positions shifted. For example, in the preferred embodiment, the volume increment value is shifted right three positions, thereby dividing it by three and effectively reducing the increment value. When the synthesizer is programmed to increment the volume at a slow rate, preferably the shift circuitry is automatically enabled. Those volume increment bits which are shifted right may be added to the current value of the volume to provide more resolution to this value.

Abstract: A karaoke apparatus is provided with a pickup device that collects a singing voice sound to convert the same into a corresponding vocal signal. Further, a music generator device generates a music signal representative of a karaoke accompaniment sound which is selected to accompany the singing voice sound. A harmony generator device generates a harmony signal representative of a harmony chorus sound which is made consonant with the singing voice sound. A first processor device processes the vocal signal to impart an effect to the collected singing voice sound. A second processor device processes the harmony signal separately from the vocal signal to impart another effect to the harmony chorus sound independently from the singing voice sound. A sound device amplifies the processed vocal signal, the processed harmony signal and the generated music signal so as to sound the singing voice sound, the harmony chorus sound and the karaoke accompaniment sound concurrently with each other.

Abstract: A stereophonic sound field generation device includes a left side localized reflected sound generation circuit which generates a left channel reflected sound signal and a left channel cancel signal for realizing an outside-of-speaker localization of the left channel reflected sound signal. The left side localized reflected sound generation circuit includes a delay circuit and a coefficient generator. By variously setting delay time of the delay circuit 140 and multiplication coefficient of the coefficient generator with respect to each reflected sound, at least one reflected sound in a broad listenable range can be localized in a space outside of a left channel speaker even when a listener moves his listening position.

Abstract: A sound reproduction control apparatus processes an audio signal in reproduction of a musical sound to control a tone of the musical sound while imparting an effect to the musical sound. A preceding tone controller has a variable frequency response for amplifying the audio signal by the variable frequency response so as to variably control the tone of the musical sound. A signal processor is connected to the preceding tone controller for processing the audio signal to impart a desired effect to the musical sound. A succeeding tone controller is connected to the processor means and has an adjustable frequency response adjusted separately from the variable frequency response of the preceding tone controller for amplifying the audio signal by the adjusted frequency response so as to adjustively control the tone of the musical sound.

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.

Abstract: A sound field control device includes a surround decoder for decoding an input surround encode signal and generating three channel signals of L, C and R to be reproduced in front left, front center and front right and one channel signal of S to be reproduced in the rear, output terminals for outputting the three channel signals, a first reflected sound generation section generating a reflected sound of the signal S, a second reflected sound generation section generating another reflected sound signal of the signal S, a first adder adding the reflected sound signal generated by the first reflected sound generation section and the signal S together, a phase shifting section shifting phase of the reflected sound signal generated by the second reflected sound generation section according to frequency of the reflected sound signal, a second adder adding a reflected sound signal provided by the phase shifting section and the signal S together, and output terminals for outputting signals of RL and RR channels for re

Abstract: System and method for enhancing the spatial effect of sound produced by a sound system. In an exemplary embodiment, the system includes a reverberator and a sound spatialization unit, which are a combination of filter(s), attenuator(s), differentiator(s), adder(s) and phase shifter(s). The present invention creates sound images at different spatial locations for different frequencies by employing a phase shifted high frequency reverberated signal. As a result, the loud speakers produce sound images located at several spatial positions, producing a perception that there is an array of loudspeakers surrounding a listener.

Abstract: There are provided a tone source for generating tone signals in plural channels independently and effect impartment sections, provided in corresponding relations to the channels, for imparting individual effects to the tone signals of the channels generated by the tone source. Each of the effect impartment sections, in accordance with tone control information unique to the tone signal of the corresponding channel, controls a parameter of the effect to be imparted. In a case where any of the effect impartment sections uses a signal delaying memory to impart an effect, when a damping (or truncating process) is performed in any of the channels, the storage area to be used for the channel is switched to another unused storage area so as to prevent any preceding tone's delayed signal from being undesirably mixed into the signal of a new tone.

Abstract: An external memory control circuit for a sound field processing digital signal processor capable of obtaining continuous sound listening by shortening the time that sound is cut off upon a change of the sound field algorithm depending on the listening place (i.e., concert hall, meeting hall, church or the like). The external memory control circuit is constructed to utilize a mute function only for the time required to transfer the algorithm itself where a sound field processing circuit is realized using a DSP, thereby reducing the time that sound is cut off and thereby obtaining continuous sound listening even when the sound field algorithm is changed while listening.

Abstract: Early reflection sounds and subsequent reverberation sounds are processed in different modes by separate processing devices in order to obtain reverberation effects close to those of natural sound as well as to freely produce reverberation effects that do not exist in natural sound. When the early reflection sound overlaps the subsequent reverberation sound in time, the early reflection sound and the subsequent reverberation sound are produced without noise. There is no need to omit either one of them. Besides, complex envelope control is executed for the early reflection sound, and the envelope levels are weighted and fluctuated. Therefore, it is possible to establish a highly sophisticated sound field.

Abstract: Small discontinuities in digital sound data caused by a pitch control operation are processed by using a secondary FIR filter and smoothing the discontinuities. When lowering the key, the data value subject to the pitch down operation is moderately approximated to that of one sampling period before over a cycle of small discontinuity occurrence. when raising the key, the data value subject to the pitch up operation is moderately approximated to the current data value over a cycle of small discontinuity occurrence. By this method, the extent of the data discontinuity is greatly reduced. Large discontinuities for every segment are also processed by cross fading two converted data sequences which are produced from the input sound data and have different timings of large discontinuity occurrence, one data value being increased from its discontinuity point while the other data value is decreased from the same point in time.

Abstract: An AC-3 decoder output signals of five channels, i.e., main signals of left front, center front and right front channels and surround signals of two channels. A first synthesizing section synthesizes the three-channel main signals and a difference signal between the two-channel surround signals, and the resultant synthesized signals are then fed to a main sound field signal forming section which in turn forms main sound field signals. A second synthesizing section synthesizes the surround signals of the two channels and a difference signal between the main signals of the left and right channels, and the resultant synthesized signals are fed to a surround sound field signal forming section which in turn forms surround sound field signals. A composite sound field forming section synthesizes the main sound field signals, surround sound field signals and original surround signals so as to form composite sound field signals.

Abstract: In a power amplifier, a control device employing a dual ganged potentiometer is connected in the feedback circuits of successive amplifier stages to increase the gain of the front end stage while simultaneously reducing the gain in a subsequent stage when the control is turned down. A distortion circuit, such as a tube compression circuit, is coupled between the stages. When the control is turned down, the compression circuit is activated at a lower level that effectively reduces the output power of the second stage. Presence and resonance controls may be provided to tailor high and low frequency damping factor of the second stage and function until the compression circuitry is activated. A ripple reduction filter is provided in a filtered supply for the drivers in the second or power amplifier stage.

Abstract: The present invention is concerned with an audio equipment capable of processing video signals and a method of displaying operation of the acoustic equipment. An input selector switch 1 is supplied with reproduced audio signals from devices connected to a tuner unit 2 and input terminals 3, and supplies a signal to a sound field processing DSP unit 4. A signal processed by the DSP unit 4 is applied through a muting switch 5 to an output terminal 6. An input selector switch 7 is supplied with reproduced video signals from devices connected to input terminals 8, and supplies a signal to a graphic controller (GDC) 9 which generates a video signal to display an image. One of a signal generated by the GDC 9 and an original video signal are selected by a switch 10, and applied to an output terminal 11. The GDC 9 also generates an image simulating a control panel. When a key switch 13 is operated, the display corresponding to the operated key in the simulating image is varied.

Abstract: A sound image enhancement apparatus for reproducing two-channel stereo signals with speakers, includes for each channel a first phase shifter and a second phase shifter for introducing different amounts of phase shift to the signals. These phase shifters may be connected in parallel or in series. This arrangement enables virtual speakers to be located at the back of a listener. An inexpensive DSP is usable, and the number of processing steps is reduced to about one third of the number when an FIR filter is used. Moreover, it is possible to reproduce reverberation sounds from the front, back and sides, thereby simulating sound fields at a live performance.

Abstract: This reverberant characteristic signal generation apparatus comprises a measuring room where a speaker and one channel of microphone is provided to supply one channel of a pulse train indicative of the reverberant characteristic of the room, a direction data generation portion for generating direction data indicative of an imaginary incoming direction of a indirectly transmitted impulse sound, an operation portion for operating the time differences due to the inclination of the incoming direction of the indirectly transmitted impulse sound to an imaginary dummy head, having right and left ears having a distance therebetween, at the position of the microphone, and an output portion for outputting an amplitude of each pulse and the delay time of the indirectly transmitted impulse sound to the microphone and the time differences to provide the reverberant characteristic signal which may be recorded by a recorder.

Abstract: A sound field control device includes an acoustic feedback signal path including a chamber and one or more loudspeaker or loudspeakers and one or more microphone or microphones provided in the chamber, the loudspeakers being disposed at a predetermined distance from the microphones, and at least either the loudspeakers or the microphones being provided in the plural, an electrical feedback signal path for feeding back a collected sound signal collected by the microphone or microphones to the loudspeaker or loudspeakers, and a collected sound signal supply control circuit provided in the electrical feedback signal path for changing, with lapse of time, a signal level of each collected sound signal fed back from the microphones to the loudspeakers. The transmission route of the collected sound signal is completely switched or the ratio of distribution of the collected sound signal is changed. Coloration in hearing is thereby reduced and a margin of howling is expanded without causing unnaturalness in hearing.

Abstract: A system for mixing five channel sound which surrounds an audio plane. The position of a sound source is displayed on a VDU (16) relative to the position of a notional listener (31). The sound source is moved within the audio plane by operation of a stylus (23) upon a touch tablet (24). Thus, an operator is only required to specify positions of a sound source over time, whereafter a processing unit (17) calculates actual gain values for the five channels at sample rate. Gain values are calculated for the sound track for each of the loudspeaker channels and for each of these specified points. Gain values are then produced at sample rate by interpolating calculated gain values for each channel at sample rate.

Abstract: A reverberation effect imparting system includes plural comb filters, each of which has a signal delay line and a feedback loop for filtering a delayed output signal from the delay line and feeding the filtered signal back to the input side with a variable loop gain. The device further includes a junction section which receives the respective output signals from the delay lines of the plural comb filters. The junction section controls the received signals with respective variable scattering coefficients and sums up the thus-controlled signals, so as to introduce the summed signal into the feedback loop of each of the comb filters. In this manner, reverberation control according to the waveguide theory is performed by the junction section, while the conventional-type reverberation control is performed by the comb filters. Thus, reverberation can be controlled through a combination of the two controls.

Abstract: A reverberation imparting apparatus comprises at least a data shift register, a coefficient shift register and an arithmetic convolution unit. The data shift register stores a string of sampling data each representative of an instantaneous value of an audio signal at each sampling period, while the coefficient shift register stores a string of coefficients which are created on the basis of waveforms of reflected sounds in response to an impulse sound artificially produced in a certain sound-field space such as a church. Normally, the data shift register successively stores new sampling data which are newly inputted thereto so that the contents of the data shift register is successively renewed. The arithmetic convolution unit performs an arithmetic convolution, using the string of coefficients on the string of sampling data.

Abstract: A sound reproduction system for automatically conducting a tone control for an audio signal according to the present invention includes: an input terminal for inputting an audio signal; a judgment circuit for determining a ratio between a stereophonic signal component and a monophonic signal component of the input audio signal and for outputting a judgment signal; a signal processing section for receiving the input audio signal and processing the input audio signal so as to generate an output signal having uniform transmission characteristics irrespective of the listening position; an adder for receiving the input audio signal and the output signal and for adding the input audio signal with the output signal at a certain addition ratio based on the judgment signal so as to generate an added signal; and a loudspeaker for receiving the added signal and for reproducing the added signal in a plurality of positions.

Abstract: A signal processor executes a plurality of microprograms to perform delaying processing and various arithmetic computation processings of digital signals input thereto. The signal processor has a storage area divided into a plurality of divided areas corresponding, respectively, to the microprograms. The write and read of the divided areas are controlled. When an instruction for changing at least one of the microprograms is given, the write and read of the digital signals are controlled such that at least one of the digital signals stored in at least one of the divided areas corresponds to the at least one microprogram is cleared without clearing the others of the digital signals stored in the others of the divided areas.

Abstract: First and second processing circuits are provided for carrying out a reverberation process of an input signal, and first and second filters are provided for applying amplitude characteristics to output signals of the first and second processing circuits. A first adder is provided for adding an output signal of the first processing circuit with an output signal of the second filter at opposed phase, and a second adder is provided for adding an output signal of the first filter with an output signal of second processing circuit in-phase. First and second speakers are provided to receive output signals of the first and second adders. The first and second amplitude characteristics are determined in accordance with a correlation coefficient of sound pressures of sounds from the first and second speakers.

Abstract: A plurality of reverberation signal generating circuits are provided for generating a plurality of reverberation signals from input audio signals of two channels. Each of the reverberation signals has a predetermined function of correlation. A pair of filters are provided for controlling the function of correlation of the reverberation signal and for applying controlled reverberation signals to reverberation speakers, thereby providing a sound field.

Abstract: Spatialization of soundfields is accomplished by filtering audio signals using filters having unvarying frequency response characteristics and amplifying signals using amplifier gains adapted in response to signals representing sound source location and/or listener position. The filters are derived using a singular value decomposition process which finds the best set of component impulse responses to approximate a given set of head related transfer functions. Efficient implementations for rendering reflection effects, and for spatializing multiple sound sources and/or generating multiple output signals are disclosed.

Abstract: According the present invention, a sound field controller for reproducing a sound field with presence comprising; input terminals for inputting an audio signal having a first and a second channel signals, a signal extracting circuit for receiving and processing the audio signal, and producing an extracted signal of the audio signal, an operation circuit for receiving the extracted signal from the signal extracting circuit, performing the convolution on the extracted signal, and generating a convolution sum signal, a delay circuit for delaying the convolution sum signal by a predetermined time, and producing a delayed signal, an adding circuit for receiving the audio signal and the delayed signal, and adding the audio signal and the delayed signal with a predetermined summation ratio to produce a summed signal, and output terminals for reproducing the summed signal to localize a sound image in a desirable direction.

Abstract: The invention relates to a device to be used for changing the acoustic properties of a room, comprising a pressure transducer (1), a loudspeaker (2) and an electronic control circuit (3) adapted to generate, in response to pressure changes in the room, an electrical signal that serves to cancel the pressure change detected by the pressure transducer (1) when it is reproduced through the loudspeaker. For improving the properties of the room also with respect to reverberation, the device of the invention further includes an ambience generator (4) adapted to receive an electrical signal proportional to the sound present in the room and to generate, in response to said signal, a signal which produces an acoustic field containing early reflections and reverberation in the room when reproduced through the loudspeaker (2).

Abstract: An audio signal processing system produces an output 24 having an illusory distance effect for a sound source signal S by feeding it via a direct signal path 25 and an indirect signal path 22, 23 passing through early reflection simulation apparatus 1 which feed an output mixing mechanism 9. A control system adjusts the relative delays 3, 4 and relative gains 5, 6 in the direct 25 and indirect 22, 23 signal paths to modify the illusory distance effect so as to substantially maintain the mathematical relationship between the gains and time delays of simulated reflections relative to first sound arrivals at the output 24 encountered for sounds at that source distance in actual rooms. Signal paths 22, 23, 24, 25 may be stereophonic or multichannel using matrix gain coefficients in the early reflection simulator 1, and may produce different simulated distances for different sound positions. A plurality of sound sources S may have different simulated distances while feeding a common early reflection simulator 1.