Abstract: The present invention provides a means for two or more remotely-located individuals to communicate information about the spatial coordinates of a location of mutual interest in a more rapid, robust, and intuitive manner than is possible with any current voice communication system.

Abstract: A computer-implemented method is disclosed for enhancing quality of an audio source. The method comprises receiving control information; receiving an initial signal from the audio source; and generating a dynamic control signal based on the control information. The control information includes attack, release, length, and gain parameters.

Abstract: Processing an audio signal associated with a sound recording made available to be rendered to an end user is disclosed. The audio signal is received. A high-order perceptual attribute of the audio signal as rendered is changed by modifying the audio signal. The modification may be based on real-time analysis of the audio signal.

Abstract: For use with an interactive display system having a generally horizontal interactive display surface, four speakers that are disposed at spaced-apart locations within a housing of the system so that a sound field produced by one or more energized speakers is directed outwardly as desired. A personal computer (PC) selectively energizes one or more of the speakers in accord with a predefined criteria so that a direction of the sound field produced is generally consistent with the disposition of a user, or a state of a software application, or a disposition of a virtual object on the interactive display surface. In one embodiment, each of the speakers is mounted in a different side or end of the interactive display system. In another embodiment, the speakers are mounted at the corners of the housing. One or more of the speakers are energized at a time to produce the desired sound field.

Abstract: A sound device is provided with an input terminal for electronic signal input; a low pass filter adapted to filter electronic signals input to the input terminal and to output low frequency range electronic signals, a switch arrangement adapted to switch between full range mode and sub woofer mode, a power amplifier adapted to amplify (1) electronic signals input from the input terminal when the switch arrangement selects full range mode, and (2) electronic signals output from the low pass filter when the switch arrangement selects sub woofer mode, and a speaker adapted to be operated by electronic signals amplified by the power amplifier.

Abstract: Preserving an audio signal in an audio system includes selecting a first audio signal from a plurality of audio signals. The first audio signal is applied to a first transducer. Mix a portion of the first audio signal with a second audio signal from the plurality of audio signals to provide a mixed audio signal. A gain of the first audio signal that is applied to the first transducer is decreased while a portion of the mixed audio signal is applied to a second transducer to preserve at least a portion of the first audio signal.

Abstract: A method of producing more than two different electric time sound signals (C(t), GF(t), DF(t), GA(t), DA(t)) from two initial electric time signals (GI(t), DI(t)). The method includes in the frequency domain, producing a central electric frequency sound signal (C(?)) from the in-phase frequency components of the initial signals; and producing two front signals (GF(t), DF(t)) by subtracting the central signal (C(t)) from the initial signals (GI(t), DI(t)) In addition, two rear signals (GA(t), DA(t)) can be produced from the out-of-phase frequency components of the initial signals. In this way, the method can be used to transform a stereophonic signal into a type 5.1 signal having five different sound signals.

Abstract: An audio processor, apparatus, and method use physical speakers to emulate one or more additional speakers. The physical speakers produce sounds that, from a listener's perspective, appear to come from at least one direction where a physical speaker is not present. Any number of additional speakers can be virtualized, such as three or five speakers that allow two speakers to emulate a 5.1 audio system.

Abstract: A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a wireless link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links.

Abstract: A sound image control apparatus and sound image control method capable of increasing the accuracy, in particular, in the distance direction while keeping fixed intervals of transmission characteristic filters, without increasing the number of the transmission characteristic filters. Sound image control apparatus 100 is provided with distance gain control section 101 that performs gain control of the sound pressure of an input monaural audio signal of one channel in the distance direction from a head to a sound image before input to virtual sound field 1 sound image localization section 120 and virtual sound field 2 sound image localization section 130, and reflected sound gain control section 102 that performs gain control by distance of a reflected sound of the sound pressure of the input monaural audio signal of one channel before input to virtual reflected sound generating section 150 that controls the sound image to be localized outside the head when the sound is listened to with headphones.

Abstract: A method for estimating the location of a sound source. First, a first microphone and a second microphone are moved relative to the sound source. Then the Intra-aural Time Difference (ITD) between the two microphones is measured. The moving and the measuring are repeated until the Intra-aural Time Difference (ITD) is smaller than a predefined threshold.

Abstract: A crosstalk cancellation system for preserving quality of sound and the parameter determining method thereof. The system can cancel a two-channel audio signal without compensating a frequency response of an ipsilateral HRTF. The method includes providing an audio signal input device for inputting the two-channel audio signal, modeling a relation between parameters of the crosstalk cancellation system and a head related transfer function (HRTF) to thereby obtain an output transfer impulse response, setting contralateral parts of the output transfer impulse response approximate to zero, setting ipsilateral parts of the output transfer impulse response equal to a delay of ipsilateral head related impulse responses (HRIRs) corresponding to the HRTF, and performing an inverse operation to compute the parameters of the crosstalk cancellation system.

Abstract: A sound reproduction apparatus capable of providing a target sensation of sound localization to a listener by using a standard head-related transfer function is provided. In a microphone amplifying section, only the high frequency components of a left sound pickup signal and a right sound pickup signal, which are input from a dummy head microphone, are delayed by a delay circuit. Consequently, since the reproduction sound of the low frequency components having small individual differences can be output earlier, a listener in a reproduction sound field space can perceive the sensation of sound localization by the reproduction sound of the low frequency components that arrive earlier. As a result, even when a standard head-related transfer function is used, it becomes possible to enable the listener in the reproduction sound field space to perceive the target sensation of sound localization.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A pulse allocating method capable of coding stereophonic voice signals efficiently. In the fixed code note retrievals of this pulse allocating method, for individual subframes, the stereophonic voice signals are compared to judge similarity between channels, and are judged on their characteristics. On the basis of the similarity between the channels and the characteristics of the stereophonic signals, the pulse numbers to be allocated to the individual channels are determined. Pulse retrievals are executed to determine the pulse positions for the individual channels, so that the pulses determined are coded.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A system and a methods 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 sound signal processing method includes calculating a difference between every few ones of input multiple channel sound signals to obtain a plurality of characteristic quantities each indicating the difference, selecting a weighting factor from a weighting factor dictionary containing a plurality of weighting factors of a plurality of channels corresponding to the characteristic quantities, weighting the sound signals by using the selected weighting factor, and adding the weighted input sound signals to generate an output sound signal.

Abstract: An apparatus for decoding an audio signal and method thereof are disclosed. The present invention includes receiving the audio signal and spatial information, identifying a type of modified spatial information, generating the modified spatial information using the spatial information, and decoding the audio signal using the modified spatial information, wherein the type of the modified spatial information includes at least one of partial spatial information, combined spatial information and expanded spatial information. Accordingly, an audio signal can be decoded into a configuration different from a configuration decided by an encoding apparatus. Even if the number of speakers is smaller or greater than that of multi-channels before execution of downmixing, it is able to generate output channels having the number equal to that of the speakers from a downmix audio signal.

Abstract: A playback apparatus includes a forming section which, on the basis of an audio signal to be played back, forms audio signals on a plurality of channels for emitting sounds from a pair of sound sources, and a signal processing section which, on each of the audio signals formed by the forming section, performs signal processing for forming a targeted sound field. The signal processing section inclines a sound pressure distribution so that, for each sound source, sound pressure levels of sounds emitted from the sound source to a listening position increase in inverse proportion to angles formed between emitting directions of the sounds emitted from the sound source to the listening position and a straight line connecting the pair of sound sources.

Abstract: A multi-channel audio system including first combining circuitry, for combining a first spectral band of a first plurality channels to provide a first bass audio signal stream; second combining circuitry, for combining the first spectral band of a second plurality channels to provide a second bass audio signal stream; and third combining circuitry, for combining a second spectral band, the second spectral band including lower frequencies than the first spectral band, of the first plurality of channels and the second plurality of channels to provide a third bass audio signal stream.

Abstract: A speaker array system for outputting, with a simplified construction, sounds audible with a nearly uniform sound volume within an arbitrarily set area. The speaker array system includes a speaker array having speaker units, a delay unit for supplying the speaker units with delayed audio signals generated by adding delays to an input audio signal, an input unit for inputting area information representing a target area to which an acoustic service is provided using an acoustic beam emitted from the speaker array and which has a normal line extending in a direction different from a normal direction of a speaker surface of the speaker array, and a control unit for calculating delays to be applied to the delay unit. Based on the delays, acoustic waves output from adjacent ones of the speaker units toward the target area are made coincident with each other and an envelope is more distorted from a spherical surface so as to face the target area as the envelope propagates closer to the target area.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. A method is provided for processing enhancing bass effect in audio signals. The method also results in the bass enhancement being computationally less intensive. The bass effect enhancement techniques are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on response of an exponential transfer function.

Abstract: An apparatus for providing data for wave field synthesis rendering in a wave field synthesis system with plurality of renderer modules, at least one loudspeaker being associated with each renderer module, and the loudspeakers associated with the renderer modules being attachable at different positions in a reproduction room, includes a provider for providing a plurality of audio files, wherein a virtual source at a source position is associated with an audio file, and a data output for providing the audio file to a renderer with which an active loudspeaker is associated, with the data output further formed to not provide the audio file to a renderer if all loudspeakers associated with the renderer are not to be active for the reproduction of the source. Thus, unnecessary data transmissions in the wave field synthesis system are avoided, while making optimum use of the renderer maximum capacity in a multi-renderer system.

Abstract: There is provided a method and a system for presenting audio signals as vibrotactile stimuli to the body in accordance with a Model Human Cochlea (MHC). Audio signals are obtained for presentation. The audio signals are separated into multiple bands of discrete frequency ranges that encompass the complete audio signal. Those signals are output to multiple vibrotactile devices. The vibrotactile devices may be positioned in a respective housing to intensify and constrain a vibrational energy from the vibrotactile devices. Output of the vibrotactile devices stimulate the cutaneous receptors of the skin at the locations where the vibrotactile devices are placed. In one embodiment, a system implements this method using audio voice coils that are embedded in a chair, which make contact with the human body while seated.

Abstract: A wide mono sound reproducing method and apparatus to widen mono sound by using 2 channel speakers. The method include separating an input mono sound signal into a plurality of decorrelated signals, generating virtual sound sources by localizing each of the separated signals at virtual locations asymmetrical about a center of a front side of a listening point by applying different head related transfer functions to the separated signals, and canceling crosstalk of the generated virtual sound sources.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: A speaker system to control directivity of a speaker unit using a plurality of microphones, and a method thereof. The method includes sensing through a plurality of channels a shock sound with an impulse pattern generated at a listening position and measuring delay values between signals of the channels, reading a predetermined listening position compensation filter coefficient in accordance with the measured delay values, and controlling directivity of the speaker unit by granting the read compensation filter coefficient on input audio signals.

Abstract: A sound-field correcting apparatus includes a sound-field correcting unit for executing, based on a correcting information, predetermined audio signal processing for correcting a sound field, an information acquiring unit for acquiring the correcting information on each position, a designating unit for designating, in a predetermined space range including the positions, a target position serving as a position at which sound-field correction is to be performed, a correcting information acquiring unit for acquiring, based on the correcting information on each position, correcting information corresponding to the target position designated by the designating unit, and a control unit for performing control based on the correcting information acquired by the correcting information acquiring unit so that the sound-field correcting unit executes the audio signal processing.

Abstract: A method of encoding multi-channel signals having two or more channels into a first signal and a second signal, and an apparatus to perform the method, the method including generating the first signal by performing a first operation using a first channel signal in the multi-channel signals; and generating the second signal by performing a second operation using a combination of the first channel signal and a second channel signal in the multi-channel signals.

Abstract: A virtual speaker apparatus and a virtual speaker processing method is disclosed. The virtual speaker apparatus uses the closest surround speaker when a virtual speaker to be virtually generated is a back left virtual speaker or a back right virtual speaker.

Abstract: A sound image localization apparatus comprises an L direct output section that produces an output signal by inputting an audio signal of a rear left audio input channel to a filter having a characteristic obtained by dividing RLD by LD, an L cross output section that produces an output signal by inputting the audio signal of the rear left audio input channel to a filter having a characteristic obtained by dividing RLC by LC, an R cross output section that produces an output signal by inputting an audio signal of a rear right audio input channel to a filter having a characteristic obtained by dividing RRC by RC, an R direct output section that produces an output signal by inputting the audio signal of the rear right audio input channel to a filter having a characteristic obtained by dividing RRD by RD, a first adding section that adds a difference signal between the output signal of the L direct output section and the output signal of the R cross output section to an audio signal of a front left audio input ch

Abstract: An apparatus for controlling a wave field synthesis renderer with audio objects includes a provider for providing a scene description, wherein the scene description defines a temporal sequence of audio objects in an audio scene and further includes information on the source position of a virtual source as well as on a start or an end of the virtual source. Furthermore, the audio object includes at least a reference to an audio file associated with the virtual source. The audio objects are processed by a processor, in order to generate a single output data stream for each renderer module, wherein both information on the position of the virtual source and the audio file itself are included in mutual association in this output data stream. With this, high portability on the one hand and high quality due to secure data consistency on the other hand are achieved.

Abstract: The designation of a localization position is received in accordance with an input position on a dashboard image in a localization-position reception window, and delay times for audio signals supplied to respective speakers are set so that the sound image is localized at the received localization position. The position where a sound image aurally perceived is localized is received, on the dashboard image in a localization-position confirmation window, from the user, until the user indicates that the position where the sound image aurally perceived is localized matches the localization position designated in the localization-position reception window. The relationship between the delay times and the localization position is re-estimated so as to correspond to the received localization position and the set delay times. In accordance with the estimated relationship, processing for re-setting the delay times is repeated.

Abstract: An array loudspeaker is constructed by stacking loudspeaker rows in a plurality of stages, each of the loudspeaker rows being formed by arranging loudspeaker elements in the form of a horizontal line. The loudspeaker rows are arranged in a zigzag form by, for example, offsetting the positions of the loudspeakers in the left and right direction for each stage. As for a high range signal, a two-stage portion of halves of the loudspeaker rows is used to narrow the pitch of the loudspeaker elements so as to prevent the grating lobe. As for a low range signal, the entire single-stage loudspeaker row is used to alleviate the deviation with respect to the directivity characteristics of the high range.

Abstract: A first signal processing section is used to constitute an impulse response of a head related transfer function region contributing to position recognition of a virtual sound source. A second signal processing section is used to realize an impulse response of a reflected sound region contributing to a sense of distance of the virtual sound source. A rotational angular speed sensor and a response characteristic control section are used to recognize position changes of the listener's ears. By controlling the first signal processing section, the sound image is localized with sufficient feeling of distance at an arbitrary position while suppressing signal processing operation quantity of the impulse response corresponding to the transfer function from the virtual sound source to both ears, and the position of the virtual sound source can be changed in correspondence with a change of position of the ears of the listener.

Abstract: An apparatus for effecting sound stage expansion in an audio system presenting two sound channels includes: (a) A first signal source coupled for providing at least one first signal representing a first sound channel to at least one first input locus of a first amplifying unit. The first amplifying unit participates in presenting the first sound channel. (b) A second signal source coupled for providing at least one second signal representing a second sound channel to at least one second input locus of a second amplifying unit. The second amplifying unit participates in presenting the second sound channel. (c) At least one first filter unit coupling the first signal source with at least one of the at least one second input locus. (d) At least one second filter unit coupling the second signal source with at least one of the at least one first input locus.

Abstract: Two-channel input audio signals are processed to construct output audio signals by decomposing the two-channel input audio signals into a plurality of two-channel subband audio signals. Separately, in each of a plurality of subbands, at least three generated subband audio signals are generated by steering the two-channel subband audio signals into at least three generated signal locations. The output audio signals are synthesized from the generated subband audio signals. The steering applies differing construction rules in at least two of the plurality of subbands.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A system and method for optimizing digital audio playback is disclosed. The system and method can prevent playback of a digital audio file in an enhanced mode if the digital audio file does not meet a predetermined criterion, such as a particular bit rate. One example of an enhanced mode is a simulated multi-channel surround sound mode.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: Multi-channel audio signals arranged to achieve original two-dimensional sound image localization are input, and the audio signal of each channel, included in the input multi-channel audio signals, is distributed to individual output channels. Each of the distributed signals is multiplied by a corresponding coefficient determined independently for each of the output channels, in accordance with a deviation from the original two-dimensional sound image localization. Then, the audio signals distributed to the individual output channels and multiplied by the corresponding coefficients are summed up, separately for each of the output channels. Thus, the summed-up audio signals of the individual output channels are output as multi-channel audio signals having the sound image localization varied in accordance with the deviation. If the deviation from the original two-dimensional sound image localization is varied over time, a panning effect can be achieved.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: This system for determining a representation of an acoustic field (P) includes: acoustic wave acquisition elements (1) including a plurality of elemental sensors (21 to 2Q) which are distributed in space and which each deliver a measurement signal (c1 to cQ); and elements (8) for processing by the application, to the measurement signals (c1 to cQ), of filtering combinations representative of structural characteristics of the acquisition elements (1) in order to deliver a plurality of acoustic signals (sc1 to scN) which are each associated with a predetermined general reproduction direction defined relative to a given point in space (14), the set of acoustic signals (sc1 to scN) forming a representation of the acoustic field (P). The system is characterized in that the elemental sensors (21 to 2Q) are distributed in space in a substantially non-regular manner and in that the filtering combinations are representative of that distribution.

Abstract: To realize virtual surround reproduction with small deterioration in the SN ratio of an entire reproduction system, in the case of adding together virtual surround signals and front-channel audio signals in an adder 1300, volume normalization is carried out in a volume normalizing section 1200 before the addition. In the case where the addition is not performed, the virtual surround signals and the front-channel audio signals are output independently of each other without being subjected to volume normalization in the volume normalizing section 1200.

Abstract: An audio spatial environment engine for flexible and scalable up-mixing from an M channel audio system to an N channel audio system, where M and N are integers and N is greater than M, is provided. The input M channel audio is provided to an analysis filter bank which converts the time domain signals into frequency domain signals. Relevant inter-channel spatial cues are extracted from the frequency domain signals on a sub-band basis and are used as parameters to generate adaptive N channel filters which control the spatial placement of a frequency band element in the up-mixed sound field. The N channel filters are smoothed across both time and frequency to limit filter variability which could cause annoying fluctuation effects. The smoothed N channel filters are then applied to adaptive combinations of the frequency domain input signals and are provided to a synthesis filter bank which generates the N channel time domain output signals.

Abstract: An integrated individual listening device and decoder for receiving an audio signal including a decoder for decoding the audio signal by separating the audio signal into a voice signal and a background signal, a first end-user adjustable amplifier coupled to the voice signal and amplifying the voice signal; a second end-user adjustable amplifier coupled to the background signal and amplifying the background signal; a summing amplifier coupled to outputs of said first and second end-user adjustable amplifiers and outputting a total audio signal, said total signal being coupled to an individual listening device.