Abstract: In an adaptive phase discovery system a first audio signal is received via a first microphone and a second signal is received via a second microphone. Corresponding audio frames of the first and second signals are each transformed into the frequency domain and a plurality of frequency sub-bands are generated. A phase is determined for each frequency sub-band in each signal. Instantaneous phase differences are determined between the signals at each of the frequency sub-bands. Lower frequency instantaneous phase differences are filtered over time to determine current phase differences at lower frequencies. When SNR is high in lower frequency sub-bands, lower frequency sub-band phase differences are tracked to the higher frequency sub-bands. The tracked higher frequency phase differences are filtered over time to determine phase differences for the current frame. The phase differences may be used to rotate phases in each sub-band and sum signals and/or to reject off-axis signals.

Abstract: A multi-channel audio system that can provide a variable sub-sample delay between two or more audio channels. In one embodiment, a variable timing clock generator generates multiple clock signals where each may have different phase, and the clock generator can vary the phase difference, in accordance with a sub-sample delay setting input. These clock signals are used by respective digital-to-analog converters (DACs) to convert the digital audio channels into analog form. In another embodiment, a variable delay block is added to an oversampling DAC, on a per channel basis. Other embodiments are also described and claimed.

Abstract: Zoom motor noise in a camera audio recording is reduced by detecting activity of the zoom motor, transforming a audio signal into the frequency domain during zoom motor activity, and scaling the frequency domain signal during zoom motor activity in each of a series of frequency bins by a scaling factor derived from a pre-stored zoom motor noise spectrum to produce a processed audio signal in the frequency domain. The processed audio signal is then transformed back to the time domain.

Abstract: As described herein, a sound pickup for musical cymbals includes an integrated assembly attachable to a cymbal stand. The integrated assembly includes a plurality of microphones arranged and electrically connected such that the resulting amplified sound is of optimal quality and of relatively constant loudness regardless of cymbal tilt. In one embodiment, two microphones are used, with the signal phase from one microphone being inverted prior to combination with the signal from the other microphone. The inversion is implemented using an inverter and serves to cancel signals that are in phase with one another and augment signals that are out of phase with one another. This, along with suitable placement of the pickup, exploits the fact that the more desirable components of the cymbal's vibration at the inflection point of the cymbal are out of phase with each other, whereas the less-desirable components are in phase with each other.

Abstract: There is provided a sound collecting device, including: an orientation direction forming section that forms an orientation direction of a microphone array; and a control section that, when a characteristic in a frequency band of a synthesized signal obtained by synthesizing the acoustic signals corresponds to a characteristic of an acoustic signal corresponding to a sound other than a target sound, controls the orientation direction forming section such that an orientation direction that is a direction that is different than an orientation direction of the microphone array at a present point in time is formed, and, when the characteristic in the frequency band of the synthesized signal does not correspond to a characteristic of an acoustic signal corresponding to a sound other than the target sound, controls the orientation direction forming section such that the orientation direction of the microphone array is maintained.

Abstract: Time interval sound alignment techniques are described. In one or more implementations, one or more inputs are received via interaction with a user interface that indicate that a first time interval in a first representation of sound data generated from a first sound signal corresponds to a second time interval in a second representation of sound data generated from a second sound signal. A stretch value is calculated based on an amount of time represented in the first and second time intervals, respectively. Aligned sound data is generated from the sound data for the first and second time intervals based on the calculated stretch value.

Abstract: Based on phase differences between corresponding frequency components of different channels of a multichannel signal, a measure of directional coherency is calculated. Application of such a measure to voice activity detection and noise reduction are also disclosed.

Abstract: The active noise control device includes: a signal obtaining section that obtains an electric signal relating to the predetermined sound; a control section that adjusts an amplitude and a phase of the electric signal obtained by the signal obtaining section; a vibrating section having a diaphragm and a vibrator, the vibrator vibrating in accordance with an output from the control section. Because a sound radiated from the diaphragm toward the first region is substantially in opposite phase to that toward the second region, the control section controls the vibrator so that the diaphragm generates a sound that attenuates the predetermined sound in the first region, and causes the predetermined sound to have a desired frequency characteristic in the second region.

Abstract: An audio processor for converting a multi-channel audio input signal, such as a B-format sound field signal, into a set of audio output signals, such as a set of two or more audio output signals arranged for headphone reproduction or for playback over an array of loudspeakers. A filter bank splits each of the input channels into frequency bands. The input signal is decomposed into plane waves to determine one or two dominant sound source directions. The(se) are used to determine a set of virtual loudspeaker positions selected such that the dominant direction(s) coincide(s) with virtual loudspeaker positions. The input signal is decoded into virtual loudspeaker signals corresponding to each of the virtual loudspeaker positions, and the virtual loudspeaker signals are processed with transfer functions suitable to create the illusion of sound emanating from the directions of the virtual loudspeakers.

Abstract: This invention allows changing the volume of a received specific input audio without influencing other audio output devices in a terminal apparatus to which a plurality of audio output devices is connected. This invention provides a terminal apparatus, which is connected to a plurality of audio output devices, and controls to output a plurality of audio data to any of the audio output devices, including a determination unit which determines the audio output devices used to output the plurality of audio data based on information indicating the relationship among audio input devices used to generate the plurality of audio data, and information indicating the relationship among the audio output devices, a mixing unit which mixes the plurality of audio data when it is determined that the plurality of audio data are output to one audio output device.

Abstract: A signal processor to produce a simulated reverberation effect based on an input signal and conveying the impression of multiple interconnected resonating spaces. A feedback delay network produces a reverberation tail signal, which is delayed by varying amounts in a delay module. A panning module produces a multi-channel signal based on the reverberation tail signal and its echoes.

Abstract: A method of setting a parameter for an audio apparatus including a delay processing unit which applies a delaying process and an amplifying process on an input audio signal in accordance with the set parameter, includes: performing a measuring process including: causing an outputting device to output measurement sound; and measuring a first impulse response at a sound receiving point; performing a specifying process including: analyzing the measured first impulse response; calculating a second impulse response at the sound receiving point when the outputting device outputs sound in case where a parameter is set in the audio apparatus and an audio signal indicative of the measurement sound is input to the audio apparatus; and specifying a parameter in accordance with the calculated second impulse response; and performing a setting process of setting the specified parameter in the delay processing unit.

Abstract: A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audio perception characteristics (sometimes referred to as the listener's Personal Waveform Profile) to transform audio content so that the listener perceives the content similarly to how the content would be perceived by a different listener. An audio testing tool may be implemented as software application to identify and capture respective listeners' Personal Waveform Profiles. A signal processor may operate an algorithm processing source audio content using the respective listeners' Personal Waveform Profiles to provide audio output based on a difference between different profiles.

Abstract: There is provided a sound processing apparatus for processing received sounds. A plurality of sound receiving units which are included in the apparatus output individually a sound signal corresponding to a received sound, then the sound signals in a time domain are converted into respective converted signal in a frequency domain, and a spectral ratio between the two converted signals is calculated for driving a phase correction value which corrects a phase of the sound signal.

Abstract: According to one embodiment, an amplitude modulation unit generates a first signal by modulating an amplitude of a carrier wave signal having a frequency of an ultrasonic band, based on an acoustic signal. A phase control unit generates a second signal and a third signal by controlling a phase of the first signal. Respective phases of the second signal and the third signal are approximately opposite. A first parametric loudspeaker radiates a first sound wave toward a first control point, based on the second signal. A first reflection unit has a first concave to receive a sound wave and reflect the sound wave toward the first control point. A focal point of the sound wave reflected by the first concave is the first control point. A second parametric loudspeaker radiates a second sound wave toward the first concave, based on the third signal.

Abstract: A signal processing device includes: a sound adjustment amount calculation unit which calculates a sound adjustment amount for adjusting sound characteristics of each channel to a predetermined sound characteristic for each channel, using a sound signal that is obtained by collecting the outputs of each channel; an evaluation value calculation unit which calculates a coefficient allocation evaluation value for allocating a size of a filter coefficient necessary for the sound adjustment of the respective channels for each channel, based on the sound adjustment amount that is calculated by the sound adjustment amount calculation unit; and a filter coefficient calculation unit which calculates the filter coefficient for each channel using the coefficient allocation evaluation value that is calculated by the evaluation value calculation unit.

Abstract: A display device includes a display portion that displays video, a first audio output portion that outputs stereo audio of a high frequency range, and that is a surface sound source positioned on a rear surface of the display portion, on one of an upper section and a lower section of the display portion, a second audio output portion that outputs stereo audio of a lower frequency range, and that is one of a surface sound source and a point sound source positioned on the rear surface of the display portion, on which the first audio output portion is not positioned, and a delaying portion that delays output of the second audio output portion to be later than output of the first audio output portion.

Abstract: A system, device, and method for recording audio with the character and sonic benefits of a genuine analog recording is disclosed. More specifically, an electro-mechanical-software controlled closed loop analog signal processor (“CLASP”) system which is comprised of a CLASP unit containing firmware, a latency detection module, and CLASP hardware display and controls. The CLASP system further comprises CLASP software operably running on a digital audio workstation (“DAW”) which is also in operable communication with the CLASP unit. The CLASP unit is also in operable communication with an analog recordable medium. An analog audio signal is recorded on the analog recordable medium, which may consist of a coated tape, cup, cylinder, drum, or disk, and then immediately played back and routed to the DAW via an analog to digital converter, thus providing for digitally recorded analog audio. The CLASP system may also include converters and a mixing console.

Abstract: An apparatus and a method for providing a stereo effect when playing an audio file at a portable terminal are provided. The apparatus includes a component extractor for extracting a mono component and stereo components from an audio signal and a gain controller for determining a signal ratio of the stereo components to the mono component and for controlling a stereo gain according to the signal ratio.

Abstract: A sound processing device includes: a plurality of sound input units; a detecting unit for detecting a frequency component of each sound input to the plurality of sound signal unit, the each sound arriving from a direction approximately perpendicular to a line determined by arrangement positions of two sound input units among the plurality of sound input units; a correction coefficient unit for obtaining a correction coefficient for correcting a level of at least one of the sound signals generated from the input sounds by the two sound input units so as to match the levels of the sound signals with each other based on the sound of the detected frequency component; a correcting unit for correcting the level of at least one of the sound signals using the obtained correction coefficient; and a processing unit for performing a sound process based on the sound signal with the corrected level.

Abstract: Systems and methods for reducing the effects of stray magnetic flux are provided. For example, an electronic device can employ the system and can include a first audio component configured to have a first acoustic phase and a first magnetic phase. The electronic device can also include a second audio component configured to have the first acoustic phase and a second magnetic phase that is opposite the first magnetic phase. The first audio component can be positioned with respect to the second audio component, such that any stray magnetic flux from the first audio component enters the second audio component during operation of the first and second audio components.

Abstract: A vehicle existence informing device is configured to inform existence of a vehicle by using a warning sound at a frequency in an audible range. A speaker array includes at least two speakers arranged so that oscillation directions of oscillators of the at least two speakers substantially coincide with each other. The speaker array is configured to radiate a warning sound in air on a carrier wave, which is at a frequency in an ultrasonic range. A phase control unit is configured to advance and delay a phase of a sound wave radiated from one of the at least two speakers with respect to a phase of a sound wave radiated from another of the at least two speakers.

Abstract: A sensor network system including node devices connected in a network via predetermined propagation paths collects data measured at each node device to be aggregated into one base station via a time-synchronized sensor network system. The base station calculates a position of the signal source based on the angle estimation value of the signal from each node device and position information thereof, designates a node device located nearest to the signal source as a cluster head node device, and transmits information of the position of the signal source and the designated cluster head node device to each node device, to cluster each node device located within the number of hops from the cluster head node device as a node device belonging to each cluster. Each node device performs an emphasizing process on the received signal from the signal source, and transmits an emphasized signal to the base station.

Abstract: Methods and apparatus are disclosed for subwoofer calibration. An example method includes determining a crossover frequency of a playback system, and the playback system includes the subwoofer and a playback device. The example method includes the subwoofer and the playback device configured to output a multimedia content in synchronization. The example method also includes outputting from the subwoofer and the playback device a series of tones near the crossover frequency. The example method also includes prompting a user to select a tone set from the series of tones, and includes automatically adjusting a phase of the subwoofer in relation to the playback device based on the user selection.

Abstract: Modern electronic devices are getting more portable and smaller leading to smaller distances between speakers. In particular, computers are now so compact that the notebook computer is one of the most popular computer types. However, with the proliferation of media available in digital form, both music recordings and video features, the demand for high quality reproductions on computers has increased. Systems and methods for producing wider speaker effects and immersion effects disclosed can enhance a listener's experience even in a notebook computer.

Abstract: A coefficient calculation unit calculates a coefficient (p) according to input signals (x[n]). A balance setting unit decides, according to this coefficient (p), a coefficient used to control the level of a signal in a digital level correction unit and a coefficient used to adjust the characteristics of a signal in a characteristic correction unit, thus allowing the digital level correction unit and characteristic correction unit to execute adaptive processing.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: The present invention relates to an apparatus and method for redeeming otherwise closed and concealed information contained in audio signals. An active circuit balances the ratio between in-phase and out-of-phase signals through the application of sum and difference signals and adjusts the ratio of gain in stereophonic signals as well as in monophonic and multichannel signal applications. This includes both the primary reference signal, and a plurality of redundant duplicate signals, substantially identical in all respects to the primary reference signal except in relation to magnitude and phase, for the purpose of unfolding, or opening the audio signal content. A pair of output signal levels approximates the golden ratio where the golden ratio is one plus the square root of five divided by two which gives an irrational number 1.618.

Abstract: A signal processing unit is provided. The signal processing unit includes an orthogonal transforming part including at least two sound input parts receiving input sound signals on a time axis, the orthogonal transforming part transforming two of the input sound signals into respective spectral signals on a frequency axis, a phase difference calculating part obtaining a phase difference between the two spectral signals on the frequency axis, and a filter part phasing, when the phase difference is within a given range, each component of a first one of the two spectral signals based on the phase difference at each frequency to calculate a phased spectral signal and combining the phased spectral signal and a second one of the two spectral signals to calculate a filtered spectral signal.

Abstract: A device for processing a signal includes a processing module used to process data in a domain. The device includes domain conversion hardware and memory, both separate from the processing module. The memory receives and sends data represented in a first domain to the domain conversion hardware. The domain conversion hardware converts the first domain data into the second domain data that may be used or processed by the processing module. The second domain data is then accessed and used by the processing module for processing.

Abstract: A sound system comprises at least three audio driver arrangements (101, 103, 105) for emitting audio signals. The audio driver arrangements (101, 103, 105) are angled relative to each other to emit sound signals in different directions angled at least 45° apart. A driving unit (201, 203, 205) is provided for each audio driver arrangement (101, 103, 105) to generate a drive signal. The audio driver arrangements (103, 105) angled most relative to each other are arranged to emit audio signals being between 90° and 270° out of phase. The driving arrangement (201) for the third audio driver arrangement (101) has a varying phase response with a variation resulting in the emitted audio signal varying between a first phase interval proximal the signal emitted from a first of the other audio driver arrangements (103) and a second phase interval proximal to the signal emitted from the second of the other audio driver arrangements (105). The invention may e.g.

Abstract: A range finding audio system automatically modifies the audio output of an audio source based on the distance of a listener from the speakers. A speaker in an audio system may include a range device coupled with a controller. The range device may utilize infrared, laser, or acoustic technology to determine the distance between the speaker and the listener. The controller may transfer distance information to an audio interface of a processor unit. The audio lo interface may include a positioning routine to modify the audio output according to the distance from the speaker to the listener. Alternatively, the controller may perform the functions ascribed to the positioning routine making the necessary modifications to the audio output based on the distance information.

Abstract: A sound processing device includes an inputting section which inputs L-ch audio data and R-ch audio data, a delaying section which applies a delaying process to the L-ch audio data and the R-ch audio data for a delay time that is set in a range from 62.

Abstract: Disclosed is a sound generation device including a receiver which receives a sound emission instructing signal in which time data is included and a difference calculator which calculates a difference between a timing indicated in the received time data and a timing when the sound emission instructing signal is received by the receiver, when the sound emission instructing signal in which the time data is included is received. The sound generation device further includes a histogram creator which creates a histogram on the basis of the calculated difference and a difference calculated previously, when the difference is calculated and a timing controller which controls a timing for supplying the received sound emission instructing signal to a sound emission unit on the basis of the calculated difference and a most frequent difference in the created histogram, when the difference is calculated.

Abstract: An audio system for enhancing localization of sound perceived by a listener in a listening position includes two loudspeakers and a signal processing unit. The loudspeakers are arranged distant from each other and from the listening position. The sound is transmitted from each of the loudspeakers to the listening position according to a respective transfer function. The transfer functions have different phase responses over frequency. The signal processing unit is connected upstream of the loudspeakers and receives two electrical input signals to be radiated as respective sound signals by the two loudspeakers. The signal processing unit includes a phase shifter unit that phase-shifts at least one of the electrical input signals such that a difference in phase responses is constant over a substantial portion of the human audible frequency in a frequency band.

Abstract: A method of processing an input audio signal, the method comprising forming a plurality of output audio signals from the input audio signal, wherein each output audio signal is formed by performing respective processing on the input audio signal, wherein for a first output audio signal there is a target audio equalization operation comprising a target filter twice, wherein for the first output 10 audio signal, the respective processing comprises a first audio equalization operation, the first audio equalization operation being the target audio equalization operation modified to compensate for phase shifts that correspond to zeros of the transfer function of the target audio equalization operation, wherein for each output audio signal other than the first output audio signal, the respective 15 processing comprises a compensation filter that compensates for phase shifts that correspond to poles of the transfer function of the target audio equalization operation.

Abstract: Speaker systems having simple structure, high output power, and desired directivity are disclosed. In one embodiment, a speaker system includes a first sound emission surface that emits a first sound in a first direction and a second sound emission surface that emits a second sound in a second direction that intersects with the first direction at a predetermined angle. The first and second sounds include at least sounds respectively generated from a common first signal source and different in phase from each other.

Abstract: Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme.

Abstract: An apparatus for processing a signal and method thereof are disclosed. The present invention includes receiving a downmix signal generated from a plural-channel signal, mix information and phase shift information on the plural-channel signal, upmixing the downmix signal into the plural-channel signal by applying the mix information to the downmix signal, and generating an original plural-channel signal by shifting a phase of at least one channel of the plural-channel signal based on the phase shift information. According to the present invention compensate a reconstructed original plural-channel signal using compensation information (phase shift information), thereby compensating a phase or a gain lost in the plural-channel signal reconstructed by upmixing using mix information.

Abstract: A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector ?0(f) between the associated calibration microphone signals may be calculated from their frequency spectra for the calibration position. Then, during an operating mode, a signal spectrum S of a signal to be output is calculated by multiplication of at least one of the two frequency spectra of the current microphone signals with a spectral filter function F.

Abstract: A channel divider that can suitably set a crossover frequency when a biwiring-connectable multiway speaker system including network circuits are used is provided. The channel divider includes a low-pass filter LPF and a high-pass filter HPF for dividing a band of a sound signal into a first output signal and a second output signal so as to output the signals, and a control circuit for controlling the filters. The channel divider has a sound output mode in which the control circuit sets a cutoff frequency fcL of LPF to a value higher than fc0 by about ? to 1 Oct., sets a cutoff frequency fcH of HPF to a value lower than fc0 by about ? to 1 Oct., and outputs the first output signal and the second output signal including frequency bands fcL to fcH when any crossover frequency fc0 for defining the band division is specified.

Abstract: A method (100) for matching characteristics of two or more transducer systems (202, 208). The method involving: receiving input signals from a set of said transducer systems; determining if the input signals contain a pre-defined portion of a common signal which is the same at all of said transducer systems; and balancing the characteristics of the transducer systems when it is determined that the input signals contain the pre-determined portion of the common signal.

Abstract: The invention relates to processing an audio signal to correct the non-linear frequency response of a loudspeaker. An input audio stream (1801) is received, from which an input frame (1802) is generated. The input frame is subjected to a Fourier transform thereby creating a frequency spectrum defining magnitude and phase values for a plurality of frequency bands. The frequency spectrum is adjusted by scaling the magnitude and phase values of each of the plurality of frequency bands by a magnitude and phase correction coefficient. Dither may be added. An output frame (1806) is then created by performing an inverse Fourier transform on the frequency spectrum, so as to create a portion of an output audio stream (1809).

Abstract: A method includes: detecting movement of a component of the electronic device; receiving an audio signal using a microphone of the electronic device, the audio signal including a noise signal corresponding to the component movement; and in response to detecting the component movement, combining the audio signal with a phase-inverted noise signal generated by phase inverting a recorded noise signal of the event received using the microphone of the electronic device.

Abstract: A sound reproducing apparatus is provided to include a controller configured to receive first and second sound source signals, to control magnitudes and phases of the received first and second sound source signals to be heard as if sound is reproduced at a pre-determined position within a listening space, and to output first and second control sound source signals; a first speaker configured to receive the first control sound source signal and to reproduce sound; and a second speaker configured to receive the second control sound source signal and to reproduce sound, wherein the pre-determined position is distant from a listener.

Abstract: An audio processing device includes a plurality of loudspeakers outputting audio for each band, a correction filter correcting an audio signal including a plurality of bands in accordance with characteristics of the plurality of loudspeakers, and a plurality of band division filters dividing the audio signal corrected by the correction filter into bands of the loudspeakers so that a phase difference of phase characteristics is approximately 0 degree or approximately 180 degrees. The correction filter is an inverse filter set with an impulse response based on the audio outputted for each band from the plurality of loudspeakers via the plurality of band division filters.

Abstract: A sound reproducing apparatus is provided to include a control device configured to receive first and second sound source signals, controls magnitudes and phases of the received first and second sound source signals to be heard as if sound is reproduced at a pre-determined position within a listening space, and to output first and second controlled sound source signals; a first speaker configured to receive the first controlled sound source signal and to reproduce sound; a second speaker configured to receive the second controlled sound source signal and to reproduce sound; and an input device with which a user can draw the listening space and mark the pre-determined position and the user's position in the drawn listening space, wherein the pre-determined position is distant from the user, wherein the input device is made of a touch panel and the user can draw the listening space with his/her finger or a pen and mark the pre-determined position and the user's position through the touch panel, and wherein the

Abstract: A sound processing apparatus according to the present invention acquires a test signal for measuring a standing wave state emitted in a listening room, and determines a peak position or a dip position due to a standing wave based on frequency characteristics of the test signal. Next, the sound processing apparatus emits a burst signal corresponding to the frequency of the peak position or the dip position, and acquires this signal. The sound processing apparatus calculates an increment ?P of the acquired signal, which indicates an amount of increase of a peak in the trailing edge portion corresponding to the end position of the burst signal relative to a peak in the portion corresponding to the stationary portion of the burst signal, and attenuates the frequency of the above peak position or dip position of a sound signal to be output by an attenuation depending on ?P.

Abstract: A system includes multiple speakers arranged in a speaker array configuration. The system also includes one or more filters configured to filter audio signals and generate filtered audio signals. The one or more filters are configured to operate using filter coefficients associated with a desired beam pattern to be produced by the multiple speakers. The system further includes at least one amplifier configured to amplify the filtered audio signals and provide the amplified filtered audio signals to the speakers. The one or more filters reside within or are coupled to the at least one amplifier. The system may further include a controller configured to modify at least one of the filter coefficients based on a change in the speaker configuration. The filters may operate independently of a centralized processor, and a centralized processor may not even be required to provide electronic beam forming.

Abstract: In embodiments of the present invention improved capabilities are described for determining a multi-dimensional sound signature for a location within a hypothetical space, comparing the multi-dimensional sound signature to a known multi-dimensional sound signature, and modifying the hypothetical space such that the similarity between the multi-dimensional sound signature for the location within the modified hypothetical space and the known multi-dimensional sound signature is increased.