Abstract: A system and method for speaker configuration in an audio and video housing, wherein the audio and video housing may be supported by stands located on top of a gas pump. The audio and video housing may have an audio and video source, at least one video display, and circuitry to drive the pair of speakers. The speakers may have a covering to provide protection from moisture and foreign particles. The speakers may be positioned to output sound downward directed towards the top of a gas pump. The sound may then be redirected to a first and second listening area.

Abstract: An audio jack for an audio plug for use in an electrical device, comprising a non-conductive enclosure having a front side and defining a cavity having an aperture within the front side for receiving the audio plug, the aperture having a first diameter, the cavity extending along a longitudinal axis; a conductive shielding formed from sheet metal folded over at least five sides of the enclosure including the front side, the shielding defining a shielding aperture centered on the enclosure aperture and having a second diameter larger than the first diameter; and a ground contact connected to the conductive shielding.

Abstract: A sound processor includes N (N is an integer of five or more) speakers, a sound source configured to output N sound signals an additional sound source configured to output an additional sound signal and a coefficient data input section configured to input N pieces of position information indicating respectively positions of the N speakers and N coefficients indicating respectively volumes of sounds outputted from the N speakers based on the additional sound signal a coefficient data analysis section configured to generate M (M is an integer equal to or greater than two and smaller than N) adjustment coefficients based on said N pieces of position information and the N coefficients wherein the M adjustment coefficients indicate volumes of sounds outputted from M speakers of the N speakers based on the additional sound signal.

Abstract: Acoustic echoes in communications systems are distracting and undesirable. Acoustic echoes occur in communications systems where sound produced by a speaker is picked up by a microphone in a communications system. In a stereo playback environment, echo cancellation techniques become more complicated. Echo cancellation can be performed by performing echo cancellation on a center signal, which is the sum of a left channel signal and the right channel signal, or left signal and a difference signal, which is the difference of the right channel signal and the left channel signal. The adaptation rates of the two echo cancellers meet certain constraints to prevent degeneracies in the echo cancellation system.

Abstract: An audio output device includes: a speaker unit including at least first to third speakers; and an audio signal processor configured to select either one of a first mode for outputting a stereo audio signal from the first and second speakers and a second mode for outputting a monaural audio signal from the third speaker and the first or second speaker.

Abstract: When two loudspeakers play the same signal, a “phantom center” image is produced between the speakers. However, this image differs from one produced by a real center speaker. In particular, acoustical crosstalk produces a comb-filtering effect, with cancellations that may be in the frequency range needed for the intelligibility of speech. Methods for using phase decorrelation to fill in these gaps and produce a flatter magnitude response are described, reducing coloration and potentially enhancing dialogue clarity. These methods also improve headphone compatibility and reduce the tendency of the phantom image to move toward the nearest speaker.

Abstract: An apparatus and method for generating a binaural beat for brainwave induction is provided. Upon receipt of a stereo audio signal, a decorrelator extracts a common component existing in common in both channels of the stereo audio signal, and outputs the remaining stereo components except for the extracted common component. A frequency shifter generates a common component in which the binaural beat is included, by shifting a frequency of the extracted common component. First and second mixers mix the common component in which the binaural beat is included, with the remaining stereo components.

Abstract: A directional sound system is disclosed. The directional sound system (400) comprises a plurality of equalization stages (404, 406) configured to equalize an input signal; and a transducer stage (412) configured to transmit the equalized input signal; wherein the plurality of equalization stages (404, 406) comprises a first equalization stage (404) configured to employ an approximated model of the transducer stage (412) and a second equalization stage (406) configured to compensate for differences between the approximated model of the transducer stage (412) and an actual model of the transducer stage (412).

Abstract: A wire system comprising a splitter integrated between two wire portions. The wire system can be used to transmit a signal between a signal source and a first output device connected to the wire portions. At any time, a user can choose to couple one or more additional output devices to the splitter to access the same signal being transmitted to the first output device.

Abstract: An audio reproduction system includes: a first speaker arranged near a listener and behind a head of the listener with a speaker unit being held by first holding means to make it possible to mix sounds emitted from front and rear of a vibration plate of the speaker; second and third speakers held by second holding means and arranged near the listener and on left and right of the first speaker; separating means for separating and obtaining low-frequency components and medium- and high-frequency components for left and right channels from an input audio signal; means for supplying the low-frequency components separated by the separating means to the first speaker; and means for supplying the medium- and high-frequency components for the left and right channels separated by the separating means to the second and third speakers.

Abstract: Systems, methods, and apparatus, including computer program products, for audio editing are provided. In some implementations, a method is provided. The method includes receiving audio data having a first audio channel and a second audio channel. The audio data is separated, into a plurality of blocks. An amount of misalignment is determined between the first audio channel and the second audio channel for the portion of the audio data in each block using a phase difference between the first and second audio channels for each of a plurality of frequency bands. The first and second channels are aligned using the determined misalignment.

Abstract: A speaker array is provided which can increase the orientation when reproducing front channels of a surround sound, increase the density effect, improve the narrow directivity when reproducing a stereo sound and increase the selectivity in selecting reproduction methods which match setting environments. When reproducing a stereo sound by a speaker array for reproducing a surround sound by converting a sound into a beam of sound, the speaker array is divided into a sound reproducing region for an L system and a sound reproducing region for an R system at a central portion thereof. Then, each of the reproducing regions so divided is further divided in to bands. In addition, since a high frequency reproducing region has a high directivity and a strong orientation when the sound is attempted to be reproduced simultaneously by the plurality of speakers as is described above, a reproducing region is limited to part of the reproducing regions.

Abstract: A sound signal outputting device includes a receiving section which receives signals on a plurality of channels, a band splitting section which splits the signals on the plurality of channels to produce low-frequency signals whose frequencies are lower than a predetermined frequency respectively, a separating section which separates a correlated component and uncorrelated components between predetermined channels from the low-frequency signals on the plurality of channels, an uncorrelated component outputting section which applies a first directivity to the uncorrelated components of the signals on respective channels to output applied components, and a correlated component outputting section applies a second directivity to the correlated component of the signals on respective channels to output an applied component.

Abstract: A method comprising determining at least one first parameter, wherein the first parameter is dependent on a difference between at least two audio signals; determining at least one second parameter, wherein the second parameter is dependent on at least one directional component of the at least two signals; and generating at least one ambience coefficient value dependent on the at least one first parameter and the at least one second parameter.

Abstract: A voice communication apparatus includes a communication portion that receives a plurality of frames including at least a first frame having first voice data and a second frame having second voice data subsequent to the first frame, the first voice data and the second voice data being encoded by a predetermined encoding system, a decoding portion that decodes the first voice data and the second voice data received by the communication portion, a buffer that retains the first voice data and the second voice data decoded by the decoding portion, a calculation portion that calculates an amplitude envelope based on the first voice data decoded by the decoding portion, and a controlling portion that judges whether or not the second voice data decoded by the decoding portion exceeds the amplitude envelope and corrects the second voice data that exceeds the amplitude envelope.

Abstract: A sound device includes an input terminal for electronic signal input; a line out terminal adapted to output electronic signals input on the input terminal to an external area; a delay arrangement adapted to delay electronic signals input on the input terminal for a certain time; a main output arrangement adapted to output electronic signals delayed by the delay arrangement. The delay arrangement delays signals in order to accommodate a time difference between (1) electronic signals output from an external sound device connected with the line output terminal and (2) electronic signals output from the main output arrangement. The time difference between sound that is output from the sound device and the one that is output from an external sound device that is connected with line connection terminal can be eliminated. Thus, it can prevent sound quality decrease due to time difference from happening.

Abstract: A headphone down mix signal can be efficiently derived from a parametric down mix of a multi-channel signal, when modified HRTFs (head related transfer functions) are derived from HRTFs of a multi-channel signal using a level parameter having information on a level relation between two channels of the multi-channel signals such that a modified HRTF is stronger influenced by the HRTF of a channel having a higher level than by the HRTF of a channel having a lower level. Modified HRTFs are derived within the decoding process taking into account the relative strength of the channels associated to the HRTFs. The HRTFs are thus modified such that a down mix signal of a parametric representation of a multi-channel signal can directly be used to synthesize the headphone down mix signal without the need of an intermediate full parametric multi-channel reconstruction of the parametric down mix.

Abstract: A loudspeaker, amplifier, media source, user interface, and mechanical interface are all contained in a single unit. The unit has a first portion configured to mount in a vehicle radio mounting location and a second portion configured to extend significantly outside the vehicle radio mounting location.

Abstract: An audio signal processing apparatus is provided. The audio signal processing apparatus comprises a clock generator, a processing module, an amplifying module and an output module. The clock generator is used for generating a clock signal. The processing module is coupled to the clock generator for processing the clock signal and generating a processing signal. The amplifying module is coupled to the processing module for amplifying the processing signal and generating an amplifying signal. The output module is coupled to the amplifying module for outputting the amplifying signal.

Abstract: A bandwidth extension system extends the bandwidth of an acoustic signal. By shifting a portion of the signal by a frequency value, the system generates an upper bandwidth extension signal. An extended bandwidth acoustic signal may be generated from the acoustic signal, the upper bandwidth extension signal, and/or a lower bandwidth extension signal.

Abstract: Method for outputting an audio signal to an audio output, comprising outputting a first audio signal to said audio output; providing a second audio signal; determining a point in time, wherein at said point in time said first audio signal or a derivative of said first audio signal or a derivative of said second audio signal is essentially equal to zero; switching, at said point in time, said audio output from outputting said first audio signal to outputting said second audio signal.

Abstract: An audio signal processing apparatus includes a localization direction detector that detects localization directions of two-channel input audio signals, a localization direction distribution calculator that calculates a distribution of the localization directions detected by the localization direction detector, a gain table information recording unit that records gain table information defining weights corresponding to respective localization angles, a gain generator that generates a gain corresponding to an output audio signal on the basis of the distribution calculated by the localization direction distribution calculator and the gain table information recorded in the gain table information recording unit, and a synthesizing unit that synthesizes the two-channel input audio signals using the gain generated by the gain generator.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a down-mix signal and down-mix identification information from an input bitstream, determining, based on the down-mix identification information, whether the down-mix signal is a 3D down-mix signal obtained by performing a three-dimensional (3D) rendering operation, and if the down-mix signal is not 3D down-mix signal, generating a 3D down-mix signal by performing a 3D rendering operation. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: A calculating unit (201) arranges a three-dimensional spatial body containing a listening point within a virtual space in a voice processor (200). The three-dimensional spatial body is constituted by a group of predetermined unit figures. The calculating unit (201) acquires a distribution of a region and an amount of overlap between a region occupied by the three-dimensional spatial body and a region occupied by predetermined objects. A modifying unit (202) modifies reflected sounds of sounds by the predetermined objects based on the distribution of the region of overlap and the amount of overlap acquired by the calculating unit (201). An outputting unit (203) outputs the reflected sounds modified by the modifying unit (202).

Abstract: The present invention relates to a method for adapting a hearing aid with at least one input converter, a signal processing device and an output converter by using a genetic feature of the wearer to whom the hearing aid is to be adapted. Depending on the genetic feature, at least one adaptable parameter is adapted by the signal processing device. The invention further relates to a hearing device system which can be adapted to the hearing device wearer as a function of a genetic feature.

Abstract: A surround sound outputting device includes a receiving portion which receives signals on a plurality of channels, a storing portion which stores measuring sound data representing a sound, an outputting portion which outputs a sound produced based on the signals on the plurality of channels or the measuring sound data in a controlled direction and in a beam shape, a controlling portion which controls a direction of the sound output from the outputting portion, a sound collecting portion which picks up the sound output from the outputting portion to produce picked-up sound data representing the picked-up sound, an impulse response specifying portion which specifies impulse responses in respective directions from respective sound data, a path characteristic specifying portion which specifies path distances of the paths through which the sounds output in the respective directions arrive at the sound collecting portion from the outputting portion and levels of the impulse responses based on the impulse responses

Abstract: According to one embodiment, an apparatus for presenting a moving image with sound includes an input unit, a setting unit, a main beam former unit, and an output control unit. The input unit inputs data on a moving image with sound including a moving image and a plurality of channels of sounds. The setting unit sets an arrival time difference according to a user operation, the arrival time difference being a difference in time between a plurality of channels of sounds coming from a desired direction. The main beam former unit generates a directional sound in which a sound in a direction having the arrival time difference set by the setting unit is enhanced, from the plurality of channels of sounds included in the data on the moving image with sound. The output control unit outputs the directional sound along with the moving image.

Abstract: A device (10) for enhancing a multi-channel (e.g. stereo) audio signal has a parameter adjustment unit (13) for adjusting an original parameter (?, ILD, ICC) which represents an original inter-channel property of the audio signal. The device further comprises a processing unit (11) for processing the audio signal so as to produce an enhanced audio signal having the adjusted parameter (??, ILD?, ICC?). The device allows stereo widening or other multi-channel signal enhancements without introducing artifacts.

Abstract: Electronic devices and equipment may communicate over a wired communications path. The wired communications path may include one or more wires and may be associated with a headphone cable. Data may be conveyed in the form of a digital data stream containing multiple traffic channels. The digital data stream may include superframes, each of which has multiple frames of data. The frames of data may each contain a number of data slots. Some of the slots in a superframe may be used exclusively by a particular one of the traffic channels. Boundary slots may be shared between traffic channels. Data interface circuitry may implement a data dispersion algorithm that determines the pattern in which data from each traffic channel is distributed within each boundary slot. Transmitting data interface circuitry may merge traffic channels into a single data stream. Receiving data interface circuitry may reconstruct the traffic channels.

Abstract: An arbitrarily positioned cluster of three microphones can be used for stereo input of a videoconferencing system. To produce stereo input, right and left weightings for signal inputs from each of the microphones are determined. The right and left weightings correspond to preferred directive patterns for stereo input of the system. The determined right weightings are applied to the signal inputs from each of the microphones, and the weighted inputs are summed to product the right input. The same is done for the left input using the determined left weightings. The three microphones are preferably first-order, cardioid microphone capsules spaced close together in an audio unit, where each faces radially outward at 120-degrees. The orientation of the arbitrarily positioned cluster relative to the system can be determined by directly detecting the orientation or by using stored arrangements.

Abstract: A system for processing signals is disclosed and may include a single chip having an integrated Bluetooth radio and an integrated FM radio. The single chip may include at least one processor that enables selecting from a range of FM channels, a particular frequency for one of the FM channels based on an intermediate frequency (IF). The particular frequency may be selected so that it is an integer multiple of a channel spacing between neighboring allocated FM channels within the range of FM channels, and may be offset by at most one-half the channel spacing. The at least one processor may enable determining a frequency error of the selected particular frequency for the one of the FM channels. The at least one processor may also enable determining, whether the particular frequency includes an on-frequency channel based on the determined frequency error.

Abstract: A sound measuring apparatus includes an impulse response obtaining section obtaining an impulse response, a positive transform section performing a positive transform on the impulse response obtained by the impulse response obtaining section, a filter low-pass filtering the response waveform on which the positive transform was performed by the positive transform section, a frequency characteristic obtaining section obtaining a frequency characteristic of the impulse response obtained by the impulse response obtaining section, a filter characteristic setting section setting a filter characteristic of the low-pass filter so as to be variable depending upon the frequency characteristic obtained by the frequency characteristic obtaining section, and a measurement result obtaining section obtaining a measurement result about a predetermined measurement item, based on the waveform obtained by the low-pass filter.

Abstract: Provided is a signal processing method for processing a bitstream, an information storage medium including the bitstream, an encoding apparatus, and a decoding apparatus. The signal processing method includes: receiving a bitstream including additional information; extracting first information which is information associated with extraction of the additional information and is included in at least one of additional bitstream information, a skip field, and auxiliary data bits, which are included in the bitstream; and extracting and decoding the additional information by using the first information.

Abstract: Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Abstract: A system that records audio and stores the recording is provided. The system includes first and second monitoring assemblies mounted in an earpiece that occludes and forms an acoustic seal of an ear canal. The first monitoring assembly includes an ambient sound microphone (ASM) to monitor an ambient acoustic field and produce an ASM signal. The second monitoring assembly includes an ear canal microphone (ECM) to monitor an acoustic field within the ear canal and produce an ECM signal. The system also includes a data storage device configured to act as a circular buffer for continually storing at least one of the ECM signal or the ASM signal, a further data storage device and a record-activation system. The record-activation system activates the further data storage device to record a content of the data storage device.

Abstract: A system, a device and a method for pulse width modulation is disclosed. One embodiment includes a pulse width modulation device, a first pulse width modulation mapper for pulse width modulation of a data signal, or a signal derived therefrom, and a second pulse width modulation mapper for pulse width modulation of a reference signal, or a signal derived therefrom. In one or more embodiments, the data signal e.g., can be a signal for a mono audio channel, or a signal for a stereo audio channel. The reference signal can be a signal including a constant signal level or a signal including a non-constant signal level an offset signal.

Abstract: A method of ambience extraction includes analyzing an input signal to determine the time-dependent and frequency-dependent amount of ambience in the input signal, wherein the amount of ambience is determined based on a signal model and correlation quantities computed from the input signals and wherein the ambience is extracted using a multiplicative time-frequency mask. Another method of ambience extraction includes compensating a bias in the estimation of a short-term cross-correlation coefficient. In addition, systems having various modules for implementing the above methods are disclosed.

Abstract: A digital signal processing apparatus in which a first digital filter reproduces that part of an impulse response that responds fast, and a decimation filter converts the output of a delay device of the first digital filter to a digital signal having a sampling rate of ½. The digital signal is supplied to the second digital filter that reproduces that part of the impulse response that responds slowly and outputs data representing the response characteristic of this part of the impulse response. An interpolation filter converts an input signal to a signal having the same sampling rate as the digital audio signal input to the digital signal processing apparatus, and the output signal of the interpolation filter is supplied to an adder circuit.

Abstract: An audio signal is processed to derive primary and ambient components of the signal. The signal is first transformed to generate frequency-domain subband signals. Primary and ambient components are separated by comparing frequency subband content using a complex-valued similarity metric, wherein one of the primary and ambient components is determined to be the residual after the other is identified using the similarity metric.

Abstract: Audio signals that represent a sound field with increased spatial resolution are obtained by deriving signals that represent the sound field with high-order angular terms. This is accomplished by analyzing input audio signals representing the sound field with zero-order and first-order angular terms to derive statistical characteristics of one or more angular directions of acoustic energy in the sound field. Processed signals are derived from weighted combinations of the input audio signals in which the input audio signals are weighted according to the statistical characteristics. The input audio signals and the processed signals represent the sound field as a function of angular direction with angular terms of one or more orders greater than one.

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 multi-channel decoding method includes: receiving an input signal to generate a first channel output signal and a second channel output signal, wherein the input signal is mixed with a specific clock signal; and gradually changing an amplitude of the specific clock signal from a first value to a second value when switching from a first mode corresponding to a first number of channels to a second mode corresponding to a second number of channels. Systems utilizing the method and another method further comprising calibration are also disclosed.

Abstract: An apparatus for processing an audio signal and method thereof applied to an audio playback system are disclosed. The apparatus comprises a decoder, an error-correcting circuit and an audio correcting module. The method for processing audio signals in accordance with the present invention decodes the audio signal to generate a decoded signal by the decoder. Then, the error-correcting circuit performs an error-correcting algorithm in the decoded signal to generate an error indication signal and an output audio signal. And the audio correcting module corrects the output audio signal to generate a corrected audio signal when the error indication signal indicates that the output audio signal has error.

Abstract: Vehicle audio systems including directional loudspeakers, particularly directional arrays. An exemplary audio system for a vehicle includes a plurality of audio channels. The vehicle includes a first passenger position and a second passenger position ahead of the first passenger position. The audio system includes a first directional loudspeaker positioned ahead of the first passenger position and in back of the second passenger position, constructed and arranged to radiate directionally a first audio channel so that a direction toward the first passenger position is one of a low radiation direction and a high radiation direction and so that a direction toward the second passenger position is the other of a low radiation direction and a high radiation direction.

Abstract: Rebalancing of an audio signal refers to achieving a balance of perceived loudness, typically of right and left channels, given an unbalanced input. A flexible method to automatically rebalance an audio input signal is robust against noise in extreme cases through the individual channels combined in various ways as a function of the loudness ratio between input channels.

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: A stereophonic sound reproduction system for compensating a low frequency signal and a method thereof, wherein a mono component signal for compensating low frequency signals which are attenuated when removing a crosstalk of inputted left and right signals inputted is calculated using an average value between the left and right signals, left and right compensation gains which are inversely proportional to an absolute value of a power difference value between the first left and right signals, an amplitude of the calculated mono component signal is controlled according to the left and right compensation gains, and thereafter the mono component signal with the controlled amplitude is added to the left and right signals when removing the crosstalk, whereby the left and right signals from which the crosstalk is removed and to which the mono component signal is added are outputted through left and right speakers to thus prevent distortion of the low frequency signals of original stereophonic sound with maintaining a

Abstract: A single audio service includes a plurality of audio sources, which may be classified by type. The classified audio sources having a same type may be multiplexed together to form audio streams, which may be amplified to obtain a desired audio environment.

Abstract: A method for generating a parametrically encoded audio signal, the method comprising: inputting a multi-channel audio signal comprising a plurality of audio channels; generating at least one combined signal of the plurality of audio channels; and generating one or more corresponding sets of side information including channel configuration information for controlling audio source locations in a synthesis of a binaural audio signal.

Abstract: Provided is an audio decoder which can reduce an amount of arithmetic operations while suppressing occurrence of aliasing noise. The audio decoder includes: a decoder (102) and an analysis filter bank (110) which generate, from a coded down-mixed signal, the first frequency band signal (x) corresponding to a down-mixed signal (M); a channel expansion unit (130) which converts the first frequency band signal (x) generated by the analysis filter bank (110) into output signals (y) corresponding to respective audio signals of N channels, using BC information; an synthesis filter bank (140) which performs band synthesis for the output signals (y) generate by the channel expansion unit (130) and thereby converts the output signals (y) into the respective audio signals of the N channels on a time axis; and an aliasing noise detection unit (120) which detects occurrence of aliasing noise in the first frequency band signal (x).