Abstract: The invention is directed to a display and method for conducting sound system design. The display is configured to simultaneously show a sound system model comprising both audio components and sound environment components, a plurality of parameters and a graphic representation of a calculated sound system performance attribute. In response to receiving input from the user and without any additional user intervention, the display re-displays a newly-calculated sound system performance attribute such as, for example, a graphic representation in the sound system model of a sound path associated with a user-selected reflected sound identifier. A processor also simulates, based on the sound system configuration, the sound performance that a user would expect to hear positioned in an identified listening position of the sound system model.

Abstract: The present invention provides a sound reproducing system capable of accurately analyzing a reverberation characteristic of amplified sound including an arrival direction of a reverberation component, which is more natural, and has higher realistic sensation. A surround system 100 has a speaker system 130, a signal processing apparatus 120 for recognizing a reverberation characteristic of a listening room 10 and adjusting a reverberation component of sound source to be amplified on the basis of the recognized reverberation characteristic, and a microphone array 140 constructed by a plurality of microphones M disposed in the listening room 10 and having the same characteristics, and in which distances among the microphones M are determined in advance. In the case where sound source is amplified and output from the speaker system 130 to the listening room 10, the microphone array 140 collects amplified sound in a specific listening position in the listening room 10.

Abstract: The invention provides a method and system for forming an output impulse response function. The method includes the steps of creating an initial impulse response, and dividing the impulse response into a head portion and a tail portion. The tail portion is high pass filtered, and low frequency components of the head portion are boosted. The low frequency boosted and high pass filtered respective head and tail portions are then combined into a modified output impulse response, which can then be used to spatialize an audio signal by convolving it.

Abstract: An impulse response synthesis method is carried out by a dispersion calculation process for calculating a dispersion of phase characteristics in association with a given room based on a volume of the room, a degree of sound absorption of the room, and a distance between a sound source and a receiving point arranged in the room, a noise creation process for creating a noise having the dispersion of the phase characteristics calculated in the dispersion calculation process, a trend addition process for adding a phase trend to the noise created by the noise creation process in accordance with the distance between the sound source and the receiving point and obtaining a phase characteristic of a minimum-phase component from the noise added with the phase trend, and a synthesis process for synthesizing an impulse response based on the phase characteristic of the minimum-phase component, the impulse response being used to create reverberation for the room.

Abstract: A method for reducing noise disturbance associated with an audio signal received through a microphone is provided. The method initiates with magnifying a noise disturbance of the audio signal relative to a remaining component of the audio signal. Then, a sampling rate of the audio signal is decreased. Next, an even order derivative is applied to the audio signal having the decreased sampling rate to define a detection signal. Then, the noise disturbance of the audio signal is adjusted according to a statistical average of the detection signal. A system capable of canceling disturbances associated with an audio signal, a video game controller, and an integrated circuit for reducing noise disturbances associated with an audio signal are included.

Abstract: An audio equivalent of a video zoom feature for video recording and communication applications, as well as video post production processes. The audio zoom may operate in conjunction with a video zoom feature or independently. The audio zoom may be achieved by controlling reverberation effects of a signal, controlling a gain of the signal, as well as controlling the width of a directional beam from which is used to select the particular audio component to focus on. The audio zoom may operate in response to user input, such as a user selection of a particular direction, or automatically based a current environment or other factors.

Abstract: A data hiding system is described for hiding data within an audio signal. The system can be used for watermarking, data communications, audience surveying etc. The system hides data in an audio signal by adding artificial echoes whose polarity varies with the data to be hidden. In one embodiment, each data value is represented by a positive and a negative echo having different delays. A receiver can then remove the effects of natural echoes and/or periodicities in the audio signal by differencing measurements obtained at the different delays.

Abstract: A method, an apparatus, and a software product to process a plurality of input audio signals. The apparatus accepts a plurality of input signals and includes a multi-input, multi-output reverberator arranged to generate a set of output signals including delayed reverberation components simulating the reverberations a listener is likely to hear in a listening environment. The apparatus further includes a multi-input, two-output filter accepting the outputs of the reverberator and the plurality of input terminals, providing outputs for the left and right ears, and configured to implement a set of head related transfer functions corresponding to a listening environment and a set of directions of a listener in the listening environment.

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 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: A circuit, method, and system for producing artificial ambience effect for an input audio signal, mono, stereo, or surround. The ambience effect enhances artificial reverberation, replaces artificial reverberation, or synthesizes extra audio channels, such as surround channels. The circuit may include a transient reduction module and a reverberation filter. The transient reduction module may be adapted to reduce transients in an input audio signal of one or more channels. The reverberation filter maybe adapted to receive a transient-reduced signal of one or more channels corresponding to the transient-reduced signal.

Abstract: A microphone includes a housing with a user interface configured to allow selection of a voice modification. The voice modification includes at least one of distortion, delay, reverb, auto tune, pitch, and phase. An audio to electric signal converter is at least partially enclosed in the housing and is configured to convert sound vibrations into an electric voice signal. A control module is configured to generate a signal indicative of a desired sound as a function of the modification signal and the electric voice signal.

Abstract: An audio signal processing method comprises the steps of emitting a sound at a virtual sound image location in space on the outer side of a closed surface, generating measurement-based directional transfer functions corresponding to a plurality of positions on the closed surface based on a result of measuring the sound at the plurality of respective positions on the closed surface by using a directional microphone, generating composite transfer functions corresponding to the plurality of respective positions on the closed surface by respectively adding, at a specified ratio, the measurement-based directional transfer functions and auxiliary transfer functions and generating reproduction audio signals corresponding to the plurality of respective positions on the closed surface by performing a calculation process on an input audio signal in accordance with the set of composite functions.

Abstract: A circuit, method, and system for producing artificial ambience effect for an input audio signal, mono, stereo, or surround. The ambience effect enhances artificial reverberation, replaces artificial reverberation, or synthesizes extra audio channels, such as surround channels. The circuit may include a transient reduction module and a reverberation filter. The transient reduction module may be adapted to reduce transients in an input audio signal of one or more channels. The reverberation filter maybe adapted to receive a transient-reduced signal of one or more channels corresponding to the transient-reduced signal.

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 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: The invention concerns a method for reproducing spatial impression of existing spaces in multichannel or binaural listening. It consists of following steps/phases: a) Recording of sound or impulse response of a room using multiple microphones, b) Time- and frequency-dependent processing of impulse responses or recorded sound, c) Processing of sound to multichannel loudspeaker setup in order to reproduce spatial properties of sound as they were in recording room, and (alternative to c), d) Processing of impulse response to multichannel loudspeaker setup, and convolution between rendered responses and an arbitrary monophonic sound signal to introduce the spatial properties of the measurement room to the multichannel reproduction of the arbitrary sound signal, and is applied in sound studio technology, audio broadcasting, and in audio reproduction.

Abstract: Method for processing speech signal data. A speech signal is divided into frames. Each frame is characterized by a frame number T representing a unique interval of time. Each speech signal is characterized by a power spectrum with respect to frame T and frequency band ?. A speech segment and a reverberation segment of the speech signal is determined. L filter coefficients W(k) (k=1, 2, . . . , L) respectively corresponding to L frames immediately preceding frame T are computed such that the L filter coefficients minimize a function ? that is a linear combination of sum of squares of a residual speech power in the reverberation segment and a sum of squares of a subtracted speech power in the speech segment. The computed L filter coefficients are stored within storage media of the computing apparatus.

Abstract: A method and apparatus to reproduce sound using a mono speaker, and a stereo sound system employing the same. The sound reproducing method can include: adding a first channel signal and a second channel signal of an input stereo signal or an input mono signal; sound field filtering by adding a sound field effect to the added signals; convolving the signal of the first channel with a first HRTF in a first convolution process; convolving the signal of the second channel with a second HRTF in a second convolution process; and adding the signal output from the sound field filtering and the signals output from the first and second convolution processes to produce an output signal.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: Audio loudspeaker and headphone virtualizers and cross-talk cancellers and methods use separate virtual speaker locations for different Bark frequency bands and a single reverberation filter for multi-channel virtualizer inputs.

Abstract: Methods and systems for echo modulation are described. In one embodiment, intensities of a plurality of values in multiple windows of an audio signal may be obtained. The windows may be subject to a periodic boundary condition. A plurality of echo values may be calculated for each of the respective windows. The audio signal may be altered in one or more of the windows using a windowing function and echo values. Additional methods and systems are disclosed.

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: The present invention provides a method and system for reducing power consumption of a sound processor. Aspects of the invention include providing a sound processor access to at least one register having a plurality of bits corresponding to sound operations capable of being performed by the sound processor; and allowing a host processor to write a value to at least a portion of the plurality of bits in the register during sound processing to selectively disable individual operations performed by the sound processor.

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

Abstract: An apparatus comprises a processor configured to receive a first audio signal and first location data, the first location data relating to a location of a source of the first audio signal; receive a second audio signal and second location data, the second location data relating to a location of a source of the second audio signal; receive selected location data relating to a selected location; and generate a multichannel signal in dependence on the first and second audio signals, the first and second location data and the selected location data.

Abstract: Methods and systems for simulating at least one room impulse response between two or more sound sources and two or more receivers positioned in an enclosure are provided. At least one early impulse response is generated that includes early reflections from the two or more sound sources to at least one of the receivers. At least one late impulse response is generated including a reverberation portion. The late impulse response is generated to spatially shape the reverberation portion corresponding to a spatial parameter of the enclosure. The at least one early impulse response is combined with the at least one late impulse response to form the at least one simulated room impulse response.

Abstract: A stereophonic device for headphones to which a monophonic signal or a stereophonic signal is inputted comprises an uncorrelating processing unit for reducing the correlation between two signals obtained by dividing the inputted monophonic signal into two channels or two signals constituting the inputted stereophonic signal, a reflected sound adding processing unit for adding a reflected sound, and a sound image localizing processing unit for controlling the position where a sound image is localized.

Abstract: The invention is a microphone bleed simulator which processes the audio signals recorded with one spot microphone and one room mic (or one spot mic alone) and simulates the sound that would be created in a typical studio recording environment wherein multiple spot microphones and multiple room microphones distributed throughout that environment all pick up sounds from any given sound source. Such multiple microphone contributions to the mixed sound are what imbue actual recordings with much of their characteristic sound quality and sense of realism, and what make recordings of ensemble instruments sound as though the players were performing together in one place. The system and method of the invention artificially simulates this natural microphone bleed and provides the means to create scaleable, realistic multi-channel sound mixes in formats such as stereo, 5.1 and 7.1 surround. The invention is particularly well suited to processing sampled or synthesized sound.

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 cue, for example a facial expression or hand gesture, is identified, and a device communication is filtered according to the cue.

Abstract: In an impulse response processing device, a waveform dividing part divides a waveform of an original impulse response into a plurality of blocks on a time axis. A time adjusting part performs a time adjustment to reduce a time difference between successive ones of the blocks. A waveform synthesizing part generates a waveform of a new impulse response by adding the blocks on the time axis after the time adjustment by the time adjusting part. Further, a windowing part performs a windowing process to multiply each block by a window function having a window width corresponding to a width of the block, a function value of the window function decreasing from a center to both ends of the block. The waveform synthesizing part generates the new impulse response using the blocks after the windowing process by the windowing part.

Abstract: Reverberation parameters for one or more positions of interest are derived from graphics data used for displaying a computer-generated environment. For each position of interest for which reverberation parameters are desired, environmental parameters including distances and the hardness of features in a range of interest and at points on cubemap faces are automatically determined from the graphics data. The environmental parameters are stored with the graphics data and associated with each position of interest. Upon rendering of the computer-generated environment, reverberation property set values usable by a reverberation engine are calculated or interpolated between predetermined values according to the environmental parameters.

Abstract: The invention is directed to a display and method for conducting sound system design. The display is configured to simultaneously show a sound system model comprising both audio components and sound environment components, a plurality of parameters and a graphic representation of a calculated sound system performance attribute. In response to receiving input from the user and without any additional user intervention, the display re-displays a newly-calculated sound system performance attribute such as, for example, a graphic representation in the sound system model of a sound path associated with a user-selected reflected sound identifier. A processor also simulates, based on the sound system configuration, the sound performance that a user would expect to hear positioned in an identified listening position of the sound system model.

Abstract: A series of signal processing features incorporated into its design that is both functional and decorative, to enhance convenience and performance among vocalists. A hollow, cylinder shaped body is oriented lie a conventional microphone, featuring a cardioid pickup pattern and a dynamic diaphragm, as well as a three-pin/XLR connector, a grill, and an integral pop filter. A transmitter handling the conversion of the audio signal into a radio signal and broadcasts it through an antenna, is built into the handle of the microphone, resulting in a wireless microphone that is slightly larger than a standard wired microphone. The microphone also has a shock-absorption system that reduces noise while the microphone is being handled buy the vocalist during a performance.

Abstract: A method for generating a sound giving a sensation of depth by applying, after extraction, a transfer function quadrille onto electric sound signals on the left and right thereof. The transfer functions simulate the trajectories taken by the sound associated with the electric signal to be processed in order to reach two receivers, if this sound was emitted in air. The signals that have been processed, one by one, by one of the four transfer functions of the quadrille, are combined with each other, then the sound signal thus obtained is mixed with the original electric sound signal to be processed after a temporal reset.

Abstract: Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device (100) includes: a decorrelated signal generation unit (181), and a matrix operation unit (187) and a third arithmetic unit (186). The decorrelated signal generation unit (181) generates a decorrelated signal w? indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit (187) and the third arithmetic unit (186) generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w? generated by the decorrelated signal generation unit (181), using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.

Abstract: Reverberation parameters for one or more positions of interest are derived from graphics data used for displaying a computer-generated environment. For each position of interest for which reverberation parameters are desired, environmental parameters including distances and the hardness of features in a range of interest and at points on cubemap faces are automatically determined from the graphics data. The environmental parameters are stored with the graphics data and associated with each position of interest. Upon rendering of the computer-generated environment, reverberation property set values usable by a reverberation engine are calculated or interpolated between predetermined values according to the environmental parameters.

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 device (1) for adding reverberation to an input signal (s(n)) uses a corresponding transformed input signal (S(n)), for example the Fourier transform of the signal, to generate reverberation. The device comprises circuits (3, 5, 6) for combining the transformed input signal (S(n)) with a modified and delayed transformed output signal (G(Sr(n?i))) so as to produce a current transformed output signal (Sr(n)). Inversely transforming the current transformed output signal (Sr(n)) results in an output signal (sr(n)) having reverberation.

Abstract: The present invention provides a sound field correcting system and a reverberation adjusting apparatus capable of correcting a reverberation time characteristic by an easy operation without setting a complicated parameter. A surround system 100 has a sound reproducing apparatus 120 for adjusting a reverberation component of sound on the basis of a reverberation characteristic of a listening room 10 and amplifying the sound by a speaker system 130; and a microphone 140 for collecting the amplified sound in a specific listening position in the listening room 10.

Abstract: A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

Abstract: An audio characteristic correction system adapted to an audio surround system in which a sound emitted from a directional speaker (an array speaker) is reflected on a wall surface or a sound reflection board so as to create a virtual speaker, at least one of frequency-gain characteristics, frequency-phase characteristics, and gain of an audio signal input to the directional speaker is corrected such that the sound reflected on the wall surface or the sound reflection board has desired audio characteristics at a desired listening position.

Abstract: A system and a method for correcting, simultaneously at multiple-listener positions, distortions introduced by the acoustical characteristics includes warping room responses, intelligently weighing the warped room acoustical responses to form a weighted response, a low order spectral fitting to the weighted response, forming a warped filter from the low order spectral fit, and unwarping the warped filter to form the room acoustical correction filter.

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: A sound system design/simulation system provides a more realistic simulation of an existing venue by matching a measured reverberation characteristic of the existing venue and adjusting one or more acoustic parameters characterizing the model such that a predicted reverberation characteristic substantially matches the measured reverberation characteristic.

Abstract: A portable audio recording and playback system is provided. The portable audio recording and playback system can synchronously display the corresponding lyrics or character contents, and it also can selectively record user's voice or the mixed signal of the user's voice and a background music including music and a human voice of singer during playback. Or the volume of the human voice in the background music can be increased or decreased corresponding to that of the music and be outputted or repeatedly played with changing speech speed without the tone changed for achieving the effects of on-the-go vocal and/or music accompaniment, voice recording, and language learning with changing speech speed without the tone changed.

Abstract: A distributed acoustic reverberator including a network audio connection between at least two separated locations is provided. When a sound is emitted at one of the locations, the reverberator provides a reverberation signal to that location with a procedure. An algorithm implemented by the distributed reverberator includes such a procedure for each location. The delay of the network connection is an input to the procedures. Thus, a shared and acoustically consistent space including the separated locations is provided. The reverberator may include procedures for providing a reverberation signal to one location responsive to sound at another location.

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: An engine sound processing apparatus includes an engine sound signal generating section that generates an engine sound signal indicating an engine sound generated in an engine room of a vehicle, a signal processing section that includes a reverberation effect applying portion which applies a reverberation effect to the engine sound signal and a mixing portion which mixes the engine sound signal to which the reverberation effect is not applied and the engine sound signal to which the reverberation effect is applied, a reverberation effect controlling section that instructs the reverberation effect applying portion to apply the reverberation effect to the engine sound signal, a mixing balance setting section that sets a mixing balance in the mixing portion, and a speaker that outputs a sound on the basis of the engine sound signal being output from the signal processing section.