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 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: A signal of a whole audio frequency band is input to input lines CHL and CHR having band-split digital band pass filters BFL1 to BFLn and BFR1 to BFRn, attenuators ATL1 to ATLn and ATR1 to ATRn, delay elements ZL1 to ZLn and ZR1 to ZRn, and adders 21 to 24, thereby causing speakers 25 and 26 to generate sounds. The reproduced sounds are picked up by microphones 27 and 28 placed at listening positions. The obtained pick-up signals PR and PL are passed through band-split digital band pass filters BFL1′ to BFLn′ and BFR1′ to BFRn′ which are set to the same bands as those of the band-split digital band pass filters BFL1 to BFRn, to generate test data DL1(t) to DLn(t) and DR1(t) to DRn(t). On a basis of the test data, an interaural correlation coefficient &rgr;RL is calculated.

Abstract: A reverberation generating apparatus is designed for applying a reverberation effect to an acoustic signal for output based on sampling data representative of an impulse response waveform. In the apparatus, a first convolution operating section convolutes the acoustic signal with sampling data corresponding to an initial period of the impulse response waveform so as to generate an initial acoustic signal. The initial period of the impulse response waveform is determined from a point of impulse sound emission to a point after a lapse of a specified time. A second convolution operating section convolutes the acoustic signal with sampling data corresponding to a selected period of the impulse response waveform so as to generate an additional acoustic signal. An attenuation operating section recurrently outputs the additional acoustic signal while attenuating the additional acoustic signal so as to generate a late acoustic signal.

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 method and an arrangement for reproducing a stereophonic or multi-channel audio signal in a space, particularly in the passenger compartment of a motor vehicle, is described Loudspeakers for reproducing acoustic signals are arranged at a plurality of positions in the space, and an amplitude and phase of the acoustic signals radiated from each loudspeaker are influenced so that wave fronts of the acoustic signals radiated by the loudspeakers overlap to form at least one first resulting wave front for a left channel and a second resulting wave front for a right channel of the stereophonic audio signal, so that the resulting wave fronts intersect in a listening zone within the space, and yield a stereo aural impression inside the listening zone.

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: For a left-channel input audio signal, digital filters 30LL and 30LR are provided for reproduction of impulse responses used for locating an acoustic image outside the head of a listener, and for a right-channel input audio signal, digital filters 30RL and 30RR are provided for reproduction of impulse responses for locating an acoustic image outside the listener's head. An addition circuit 7L adds output signals of the digital filters 30LL and 30RL to each other and a second addition circuit 7R adds output signals of the digital filters 30LR and 30RR to each other. Output signals of the addition circuits 7L and 7R are supplied to a headphone, whereby the reproduction acoustic images of the left-channel and right-channel input audio signals are located outside the listener's head. The sampling rate in each of the digital filters 30LL-30RR is changed depending on the section of the response time of the associated impulse response to be reproduced.

Abstract: A system for enhancing sound signals at a relatively high sonic level power input and a comparatively low signal power level output which are relatively adjusted so that the signal is enhanced when reproduced. The input includes a capacitor for creating a reverberation effect in the signal and an adjustable power level varying device for causing a second harmonic effect in the signal. When the system is used in a multichannel sound system, a low pass filter is connected between channels for causing a phantom center channel effect in the sound signal.

Abstract: A sound signal is processed in two branches (10, 20).

Abstract: High and low frequency components Cn1, Cn2 are generated by causing a center channel audio signal Cn to pass through a high-pass filter HPF and a low-pass filter LPF, and the high frequency component Cn1 is supplied to the center channel loudspeaker SPC. Also, two low frequency components Cn2L, Cn2R, which have a different phase mutually, are generated by causing the low frequency component Cn2 to pass through phase shifters 2, 3. An synthesized audio signal SL is generated by synthesizing the low frequency component Cn2L and the front-left channel audio signal FL, and then supplied to the front-left channel loudspeaker SPL. An synthesized audio signal SR is generated by synthesizing the low frequency component Cn2R and the front-right channel audio signal FR, and is supplied to the front-right channel loudspeaker SPR.

Abstract: Provides a one-terminal effector having an input terminal and an output terminal structured as one terminal by providing a circuit that makes input and output of the effector common. The one-terminal effector can avoid complex cable wiring, and can prevent degradation of sound quality. The one-terminal effector includes one input/output terminal, and an effector circuit that executes a predetermined processing on an analog audio signal that is input from the input/output terminal, and outputs the processed analog audio signal from the input/output terminal. The one-terminal effector further includes an input/output signal conversion circuit that converts the input signal and the output signal into mutually different signal forms of voltage signals or current signals.

Abstract: A method and apparatus implements a novel surround sound panning paradigm. Rather than controlling the x-y position of a perceived sound source within a linear grid, the perceived sound is characterized by specifying perceived arrival energy as a function of direction of arrival. In one embodiment, perceived sound source azimuth and width (or spatial extent) are specified, which parameters are used in a novel panning law to control each output channel. In a preferred implementation, the panning control is provided in a Plug-In application for a conventional DAW environment such as Pro Tools, which application includes an interface that provides precise control over the direction and spatial extent of audio.

Abstract: A sound processing apparatus for creating virtual sound sources in a three dimensional space includes a number of modules. These include an aural exciter module; an automated panning module; a distance control module; a delay module; an occlusion and air absorption module; a Doppler module for pitch shifting; a location processor module; and an output.

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 rear speaker and a front speaker are provided in a room. A microphone is disposed at a listening position between the front and rear speakers for picking up a first sound from the front speakers and a second sound from the rear speaker. Time lags of the first sound and the second sound is obtained from the outputs of the microphone, and the time difference between the time lags is obtained. An antiphase audio signal opposite to a reflected composite signal included in the first sound picked up by the microphone is provided by the sound field correcting circuit. The antiphase audio signal with the same amplitude is delayed based on the time difference and emitted from the rear speaker so as to cancel the reflected composite signal.

Abstract: Sound picked up by a microphone of a first sound field is reproduced by a speaker of a second sound field, and a sound picked up by a microphone of the second sound field is reproduced by the speaker of the first sound field. Sound pressure detection section detects a sound pressure of a sound present in the second (or first) sound field picked up by the microphone of the second (or first) sound field, other than a sound reproduced by the speaker of the first (or second) sound field. Other sound pressure detection section detects a sound pressure with which the sound picked up by the microphone of the second (or first) sound field and reproduced by the speaker of the first (or second) sound field is picked up by the microphone of the first (or second) sound field. Sound-pressure-difference detection section adjusts a gain of an automatic gain adjustment section in such a manner that the two detected sound pressures assume a predetermined relationship.

Abstract: To produce a three dimensional sound, an original sound recording is duplicated and the original of its duplicate are then recorded with a time delay in the range of 25 ms to 990 ms, preferably 100 to 500 ms. The resulting first echo recording is duplicated to produce a second echo recording and these are recorded with another time delay, preferably less than the first but within the aforementioned range to yield a fifth tape which has a so-called third echo imparting to the sound a three-dimensionality which cannot be achieved by conventional reverberation and echo methods.

Abstract: The present invention permits the parameters of at least one of the frequency characteristic adjusting circuit and the effect circuit in its DSP to be modified for providing the line-out jacks with an optimum music signal when its speaker (SP) switch is turned off with no headphone connected. When the headphone plug is connected with no headphone jack, the effect setting switch is assigned with a function of speaker system effect setting. When the headphone plug is connected with a corresponding headphone jack, the effect setting switch is assigned with a function of headphone effect setting. Also, upon judging that the headphone plug is connected with the headphone jack, the function of an equalizer (EQ) switch is disabled.

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: A sound apparatus is constructed for directing a sound image of a virtual sound source at a designated source point to a listener in a virtual sound field. In the sound apparatus, a database provisionally memorizes acoustic transfer characteristics of the virtual sound field in correspondence to reference source points distributed radially around a center point of the listener. Left and right filters respectively filter audio signals of left and right channels according to the acoustic transfer characteristics loaded from the database. A processor computes a leftward acoustic direction from the designated source point to a left ear of the listener, and computes a rightward acoustic direction from the designated source point to a right ear of the listener.

Abstract: Computer-aided methods and systems are proposed for centrally recording and modeling the acoustics of a closed room or partially enclosed room, wherein a room response is measured locally with a local computer by generating a sound signal with at least one acoustic source and recording the acoustic response with at least one measuring microphone. Software is downloaded to the local computer from a remote central computer, and the measured data are transmitted for further processing from the local computer to the remote central computer, optionally together with additional data required for the additional processing. The amplification factors of the acoustic source output and the microphone input can be calibrated automatically. The methods and systems can also be used to process other physical measurements.

Abstract: A reverberation device is provided for adding a reverberation signal to an audio input signal. The device consists of a late reflection generator and an early reflection generator which share common delay line memory. Sparsely selected taps are utilized to generate an early reflection signal from the input signal as it passes through the delay line. Subsequently, the delayed signal is attenuated and recirculated, and multiple randomly selected taps are utilized to generate a late reflection signal with enhanced echo density. The late reflection and early reflection signals are combined with the input signal to generate an output signal with added reverberation.

Abstract: A sound field effect control apparatus and and a sound field effect control method are provided, which are capable of enabling a listener to listen to reflected sounds close to initial reflected sounds generated in an actual acoustic space, by utilizing information reflecting the position of a sound source possessed by multichannel audio source signals. Sound field effects are applied to multichannel audio source signals that are input to a sound field effect control apparatus. The multichannel audio source signals cause sound generated from an imaginary sound source at a predetermined position to be heard by the listener when converted into sound and generated by a plurality of loudspeakers. Multichannel initial reflected sound signals corresponding to initial reflected sounds that will be heard by the listener when the sound is generated from said imaginary sound source at said predetermined position in a predetermined acoustic space, are generated from the multichannel audio source signals.

Abstract: A surround signal processing apparatus and method can realize sound image localization and have reverberation effects. In the apparatus, left and right impulse measuring sections measure left and right impulses of a head related transfer function for an input audio signal based on a number of a plurality of lattices defined in a three dimensional space, horizontal and vertical angles defined by a center of a dummy head and the plurality of lattices. Left and right convolution operators convolve left and right channel signals of input audio signal with left and right impulses of head related transfer function, respectively, in order to localize sound image for input audio signal at an objective localization position. Left and right reverberators impart first and second reverberant sounds to left and right channel signals, respectively.

Abstract: Eyewear which comprises a frame front and a first and second hinge coupled to the frame front. A first and second temple are pivotably coupled to the frame front through the first and second hinge respectively. The first and second temples are movable between a closed position and a use position. The first and second hinges bias the first and second temples toward the use position. At least a first catch is disposed on the eyewear. The catch is operable to retain the temples in the closed position in opposition to the biasing of the first and second hinges.

Abstract: A karaoke device with built-in microphone includes a main body microphone, and converts an audio signal from the microphone into audio data by an A/D converter, and writes the audio data into a ring buffer by mixing with the data already stored in the ring buffer. If an echo mode is set, a delay time constant (CD) corresponding to the echo mode is determined, and on the basis thereof, a size of the ring buffer is set. The data is read from the ring buffer, and is inputted in a sound channel. If a voice effect mode is set, a reproduction frequency constant (CF) corresponding to the voice effect mode is determined, and based thereon, an inclement value of a read pointer of the ring buffer is determined, and then, the data is read from an address indicated by the read pointer. When the read pointer reaches the delay time constant, the relevant constant is subtracted from the read pointer value.

Abstract: The present invention provides a sound-field production apparatus comprising a direct-sound-generating means for generating a direct sound from a sound signal supplied to the direct-sound-generating means and a reflected-sound-generating means 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 means. 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: An input musical sound signal is converted into a digital signal by an A/D converter (10), and then supplied to a feed back loop (22) having a digital delay circuit (16), so that an echo is produced. A distortion-adding block (28) is disposed in the feed back loop (22) to add distortion components corresponding to distortion and the like due to recording and reproducing processes of a tape-recorder-type analog echo, to the echo signal. The delay time of the digital delay circuit (16) is modulated, so that fluctuation components corresponding to, for example, wow and flutter components of a tape-recorder-type analog echo are added to the signal. The echo produced in the feed back loop (22) is returned to an analog signal by a D/A converter (24), and then added to the original sound by an adder (26). A result of the addition is then output.

Abstract: A virtual acoustic environment comprises surfaces which reflect, absorb and transmit sound. Parametrisized filters are used to represent the surfaces, and parameters defining the transfer function of the filters are presented in order to represent the parametrisized filters.

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 apparatus for continuously reproducing plural sound data has a start end/terminal end determination unit for determining the start end/terminal end of the continued respective sound data, a fade-in/fade-out unit for carrying out fade-in process at the start end of plural respective sound data and/or fade-out process at the terminal end of the same, a data output unit for continuously outputting the plural sound data which have been subjected to fade-in process and/or fade-out process, and a reproduction unit for reproducing the outputted plural sound data. In reproducing continuously the plural sound data, no noise is generated at the joint portion of the adjacent sound data.

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: In accordance with a kind of effect, an effect adding section (38) adds an original microphone signal with an effect signal and outputs the added signal, or outputs an effect signal in place of the original microphone signal. A mixing section (42) mixes the effect signal and the original microphone signal at a variable mixing ratio. The mixed signal is supplied to an echo signal generating circuit (20) to generate an echo signal. An output signal of the effect adding section (38) and the echo signal output from an echo adding section (40) are mixed by a mixing circuit (18). In the case in which both an effect and an echo are added such that the microphone signal and the effect signal are added and then output, the mixing section (42) reduces the mixing ratio of the effect signal, whereby the effect is prevented from being excessively increased.

Abstract: A music apparatus is constructed for introducing a music sound into an acoustic space, and is equipped with an acoustic feedback system for introducing a reflection sound into the acoustic space. In the music apparatus, an input device sequentially provides performance information. A sound source generates a music signal representative of the music sound in response to the performance information provided from the input device. A plurality of channels of the acoustic feedback system are provided in the music apparatus, and are spatially separate from each other.

Abstract: In a surround circuit, an audio signal or another analog signal is converted to a digital signal by an A/D conversion circuit (11) and stored in a memory (12). Subsequently, the digital signal read from the memory (12) is converted to an analog signal by a D/A conversion circuit (13). The memory (12) then functions as a delay circuit, and a delayed audio signal is obtained. By superimposing the obtained delay signal to the transmitted audio signal, a surround sound is obtained. Here, a sampling frequency of either one of the A/D conversion circuit (11) and the D/A conversion circuit (13) is changed with an elapse of time, which leads the sampling frequency of the A/D conversion circuit (11) to differ from that of the D/A conversion circuit (13), and the frequency of an output signal of the D/A conversion circuit (13) differs from that of the audio signal transmitted to the A/D conversion circuit (11). Consequently, a delay signal in which the frequency of the input audio signal is dispersed can be obtained.

Abstract: A reverberation system produces reverberant sound supposed to be generated when sound emitted from a real sound source location inside a virtual acoustic room to be simulated is reflected by virtual room partitions surrounding the same, based on a set of stored impulse responses of reverberant sound actually or virtually observed at different observation positions in the vicinity of room partitions defining an actual or virtual model acoustic room. Digital filters each convolute a source signal representing the emitted sound, using coefficient parameters representing the corresponding stored impulse response, to thereby synthesize a virtual reverberant sound signal at each of simulation positions corresponding to the observation positions.

Abstract: Early reflection and reverberation processing using a decimating filter simulates the high frequency attenuation of an actual physical and acoustical environment and advantageously reduces the memory storage and computational burden of the early reflection and reverberation processing method. A method of generating a reverberation effect in a sound signal includes decimating the sound signal in the sound signal path and forming an early reflection sound signal from the decimated sound signal. The early reflection sound signal has a reduced sample rate an attenuated high frequency components in comparison to the sound signal.

Abstract: A signal processing apparatus executes a plurality of microprograms stored in a microprogram memory device in a time-sharing manner, so as to perform arithmetic operations on a digital signal entered by a signal input device. A delay memory device that delays the digital signal. The delay memory device has a plurality of delay areas that are independently provided for the respective microprograms, such that each of the microprograms that is being executed uses a corresponding one of the plurality of delay areas so as to delay the digital signal, and a common area that can be accessed by all of the microprograms. As a result, data such as tables for use in common by the plurality of microprograms can be stored, and the stored data can be easily used by each microprogram, without making the system complicated or increasing the cost of the apparatus.

Abstract: Audio spatial localization is accomplished by utilizing input parameters representing the physical and geometrical aspects of a sound source to modify a monophonic representation of the sound or voice and generate a stereo signal which simulates the acoustical effect of the localized sound. The input parameters include location and velocity, and may also include directivity, reverberation, and other aspects. The input parameters are used to generate control parameters which control voice processing. Thus, each voice is Doppler shifted, separated into left and right channels, equalized, and one channel is delayed, according to the control parameters. In addition, the left and right channels may be separated into front and back channels, which are separately processed to simulate front and back location and motion. The stereo signals may be fed into headphones, or may be fed into a crosstalk cancellation device for use with loudspeakers.

Abstract: Techniques of making a recording of or transmitting a sound field from either multiple monaural or directional sound signals that reproduce through multiple discrete loud speakers a sound field with spatial harmonics that substantially exactly match those of the original sound field. Monaural sound sources are positioned during mastering to use contributions of all speaker channels in order to preserve the spatial harmonics. If a particular arrangement of speakers is different than what is assumed during mastering, the speaker signals are rematrixed at the home, theater or other sound reproduction location so that the spatial harmonics of the sound field reproduced by the different speaker arrangement match those of the original sound field. An alternative includes recording or transmitting directional microphone signals, or their spatial harmonic components, and then matrixing these signals at the sound reproduction location in a manner that takes into account the specific speaker arrangement.

Abstract: A karaoke amplifier is constructed for amplifying an audio signal to sound a karaoke performance while processing the audio signal according to a value of a control parameter to acoustically control the karaoke performance. An analog operating member is manually operable to set an initial analog value of the control parameter. An analog/digital converter converts the initial analog value into a corresponding initial digital value. A receiver receives correction data from an external data source. A memory memorizes the received correction data. A calculator arithmetically treats the initial digital value according to the correction data retrieved from the memory to calculate a corrected value of the control parameter. A signal processor processes the audio signal according to the corrected value of the control parameter and amplifies the processed audio signal to sound the karaoke performance which is acoustically modified according to the corrected value of the control parameter.

Abstract: A digital wavetable audio synthesizer is described. The synthesizer can generate up to 32 high-quality audio digital signals or voices, including delay-based effects, at either a 44.1 KHz sample rate or at sample rates compatible with a prior art wavetable synthesizer. The synthesizer includes an address generator which has several modes of addressing wavetable data. The address generator's addressing rate controls the pitch of the synthesizer's output signal. The synthesizer performs a 10-bit interpolation, using the wavetable data addressed by the address generator, to interpolate additional data samples. When the address generator loops through a block of data, the signal path interpolates between the data at the end and start addresses of the block of data to prevent discontinuities in the generated signal. A synthesizer volume generator, which has several modes of controlling the volume, adds envelope, right offset, left offset, and effects volume to the data.

Abstract: A digital wavetable audio synthesizer is described. The synthesizer can generate up to 32 high-quality audio digital signals or voices, including delay-based effects, at either a 44.1 KHz sample rate or at sample rates compatible with a prior art wavetable synthesizer. The synthesizer includes an address generator which has several modes of addressing wavetable data. The address generator's addressing rate controls the pitch of the synthesizer's output signal. The synthesizer performs a 10-bit interpolation, using the wavetable data addressed by the address generator, to interpolate additional data samples. When the address generator loops through a block of data, the signal path interpolates between the data at the end and start addresses of the block of data to prevent discontinuities in the generated signal. A synthesizer volume generator, which has several modes of controlling the volume, adds envelope, right offset, left offset, and effects volume to the data.

Abstract: A surround sound effect control device includes a plurality of sound signal sources each for producing a sound signal, a sound source selection signal generator for generating a sound source selection signal indicative of one sound sources, and a plurality of surround effect circuits for modifying the sound signal. A memory is provided for storing data identifying one surround effect circuit correspondingly to each of the sound sources. A first switching arrangement is provided for selecting one of the plurality of sound sources in response to the sound source selection signal, and a second switching arrangement is provided for selecting one surround effect circuit in response to the data read from the memory.

Abstract: On the basis of localization control data, a sound image localization controlling circuit reproduces input audio signals via a plurality of speakers after having applied predetermined delay-involving signal processing to the audio signals, to thereby perform sound image localization processing to localize sound images of direct sounds in a desired range including an area outside a space surrounded by the speakers. The audio signals are also supplied to a sound field controlling circuit after having been delayed by a predetermined time. The sound field controlling circuit performs operations to convolute the audio signals with reflected sound parameters so as to generate reflected sounds. The output signals of the sound image localization controlling circuit and sound field controlling circuit are fed to adders each adding together the signals of same channel. The resultant added signals are then sent to the speakers in a listening room for audible reproduction.

Abstract: Sound source data is constituted by a 14-byte sound data area SA consisting of 28 samples of 4-bit adaptive differential PCM data and a sound parameter area PA consisting of loop information LP, filter information FL and a shift amount RA.In encoding the sound source data, a CD-ROM XA encoder may be employed.4-bit ADPCM data is decoded by a BRR decoder 53, pitch-converted by a pitch conversion unit 54, and then processed with envelope transform by an envelope generator 58. After that, sound outputs of left and right channels outputted with the sound volume adjusted by volumes 60L, 60R, and 16-bit PCM data with the sound volume adjusted by a mixing volume 65, are added by an adder 62. Further, the sound data from the mixing volume 65, or an addition output produced by adding the sound data from the mixing volume 65 with the sound data from the volumes 60L, 60R, is reverberated by a reverberation processor 68.

Abstract: In an echo effector for imparting an echo effect to an audio signal based on values of a plurality of parameters, a memory is provided for memorizing an interrelationship among the plurality of the parameters which are mutually related to each other. A dial is actuated for independently setting one or more of the parameters to desired values. A microcomputer dependently sets the remaining parameters to appropriate values according to the desired values and the memorized interrelationship. Further, in a sound field effector operative according to a selected program for creating an effect simulative of a sound field in matching with a room situation under which the effector is installed, a program memory stores a plurality of programs which can be selected in terms of types of sound fields and types of room situations. A first switch is operated for designating a desired type of a sound field. A second switch is operated for designating a desired type of a room situation.

Abstract: A digital audio spatialization system that incorporates accurate synthesis of three-dimensional audio spatialization cues responsive to a desired simulated location and/or velocity of one or more emitters relative to a sound receiver. Cue synthesis may also simulate the location of one or more reflective surfaces in the receiver's simulated acoustic environment. The cue synthesis techniques are suitable for economical implementation in a personal computer add-on card.

Abstract: A music sound correcting apparatus for correcting a music sound signal entered thereinto to selectively supply the corrected music sound signal to a speaker and a headphone, is provided with a jack into which a plug of the headphone is inserted, and a detector for detecting whether or not this plug of the headphone is inserted into the jack. When the detector detects that the plug is not inserted into the jack, a frequency of the entered music sound signal is changed so as to be properly reproduced by the speaker. On the other hand, when the detector detects that this plug is inserted into the jack, the entered music sound signal is corrected so as to be properly reproduced by the headphone in such a manner that a sound image produced based upon the entered music sound signal is localized to a predetermined position. As a result, preferable music sound can be reproduced even when the speaker is used and the headphone is employed irrespective of a small amount of hardware.