Abstract: A speaker system utilizes input from a transducer that receives an acoustical signal produced by a speaker that is separate from the speaker system. The acoustical signal is amplified and drives a speaker of the speaker system. The speaker system is thereby provided with the necessary electrical audio signal without the speaker system being wired into any existing sound system wiring. In automobiles, the transducer is in proximity to a speaker wired into the automobile's audio system, and the transducer obtains the acoustical signal to produce the electrical audio signal used by the speaker of the speaker system so that no access to high or low level electrical audio signals of the audio system of the automobile is necessary. Additionally, the speaker system may employ a power socket plug that is electrically coupled to the power input of the amplifier and that may be plugged into a power socket such as those typical of most vehicles to provide electrical power to the amplifier of the speaker system.

Abstract: On an encoder-side, a multi-channel input signal is analyzed for obtaining smoothing control information, which is to be used by a decoder-side multi-channel synthesis for smoothing quantized transmitted parameters or values derived from the quantized transmitted parameters for providing an improved subjective audio quality in particular for slowly moving point sources and rapidly moving point sources having tonal material such as fast moving sinusoids.

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 method for generating and playing audio signals and a system for processing audio signals are disclosed. The method for generating audio signals includes: generating distance information about an audio signal corresponding to a view point position, according to obtained auxiliary video and direction information about the audio signal, where the auxiliary video is a disparity map or a depth map; encoding the direction information and distance information about the audio signal, and sending the encoded information. The apparatus for generating audio signals includes an audio signal distance information obtaining module and an audio signal encoding module. With the present invention, the position information, including direction information and distance information, about the audio signal may be obtained accurately in combination with a three-dimensional video signal and a three-dimensional audio signal, without increasing the size of a microphone array, and the audio signal is sent and played.

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

Abstract: 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: 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: Modifying a panned source in an audio signal comprising a plurality of channel signals is disclosed. Portions associated with the panned source are identified in at least selected ones of the channel signals. The identified portions are modified based at least in part on a user input.

Abstract: Provided is a signal processing method which calculates a correlation coefficient indicating the degree of relation in a stereo signal and extracts a speech signal from the stereo signal by using the correlation coefficient and the stereo signal.

Abstract: A transmitting/receiving method that can easily detect an influence exerted in transmission channels at a transmitter/receiver is provided. An acquiring unit 12 acquires left audio data and right audio data of audio data received by a metadata calculator 1. An adding unit 141 then calculates added data based on an added value for a predetermined time of a sum of the acquired left audio data and right audio data. Similarly, a subtracting unit 142 calculates subtracted data based on an added value for predetermined time of a difference between the acquired right audio data and left audio data. An addition unit 17 adds the added data and the subtracted data calculated by these adding unit 141 and subtracting unit 142 to the received audio data, and a transmission unit 18 transmits the audio data to which the added data and the subtracted data are added.

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: A stereo speaker system including: a speaker unit having a plurality of speakers which output audio; and a filter unit selectively receiving low frequency components of the outputted audio. The filter unit eliminates the received low frequency components to decrease back pressure on the speakers. Thus, the phenomenon of air-loading caused by back-pressure is removed and the low frequency characteristic of stereo speakers is improved by activating the vibration of a vibrating plate.

Abstract: An apparatus of generating a multi-channel sound signal is provided. The apparatus may include a sound separator to determine a number (N) of sound signals based on at least one of a mixing characteristic and a spatial characteristic of a multi-channel sound signal when receiving the multi-channel sound signal, and to separate the multi-channel sound signal into N sound signals, the sound signals being generated such that the multi-channel sound signal is separated, and a sound synthesizer to synthesize N sound signals to be M sound signals.

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: 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 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: Challenges to the implementation of equalization in the 2.1 environment arise from the constraints imposed by HD audio requirements and Windows® Vista™. A hybrid software hardware solution overcomes many of the challenges by exploiting the software capability for equalization and using a hardware codec to perform the separation into high frequency and low frequency audio streams needed to drive stereo speakers and a subwoofer.

Abstract: A method and ASIC for canceling crosstalk between a first stereo channel and a second stereo channel, wherein a first signal is input to a first output amplifier for the first channel, and a second signal is input to a second output amplifier for the second channel, and an output load for each output amplifier is connected between each output amplifier and a reference amplifier. In one embodiment, the first and second signals are split prior to inputting the signals to the first and second output amplifiers, and a gain-adjusted portion of each signal is added to the other signal on the inputs of the output amplifiers. In another embodiment, the first and second input signals are again split into two paths each. While a first path of each signal is inputted to each signal's respective output amplifier, the second paths of the first and second signals are adding together. The resulting sum is adjusted by a gain function, biased by a suitable DC voltage, and input to the reference amplifier.

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: 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: 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: Calculation is performed for sound paths 112-1, 114-1 along which sounds emitted from a sound emitting point 104 in an acoustic space 102 are reflected and delivered to a sound receiving point 106. By the calculation, entering angles eR1, eR2 by which the sound paths enter the front side 106a of the sound receiving point 106 are obtained. Calculation is then performed to obtain angles by which respective speakers 52C, 52L, 52R, 52SR, 52SL of a 5.1 surround system are arranged in a listening room, with the front side 106a of the sound receiving point 106 centered thereon. Audio signals on the respective sound paths are distributed among channels for any two speakers. Consequently, sharp localization of sound images is achieved, requiring less calculation in simulating acoustic characteristics of the acoustic space 102 in which the sound emitting point 104 for emitting sounds and the sound receiving point 106 for receiving the sounds are placed.

Abstract: Fluctuation in decoded signal localization is suppressed to maintain the feel of stereo. A selection unit (220) selects balance parameters when the balance parameters are input from a gain coefficient decoding unit (210), or selects balance parameters input from a gain coefficient calculation unit (223) when there is no balance parameter input from the gain coefficient decoding unit (210), and outputs the selected balance parameters to a multiplication unit (221). The multiplication unit (221) multiplies a gain coefficient input from the selection unit (220) with a decoded monaural signal input from a monaural decoding unit (202) to perform balance adjustment processing.

Abstract: An acoustic correction apparatus processes a pair of left and right input signals to compensate for spatial distortion as a function of frequency when said input signals are reproduced through loudspeakers in a sound system. The sound-energy of the left and right input signals is separated and corrected in a first low-frequency range and a second high-frequency range. The resultant signals are recombined to create image-corrected audio signals having a desired sound-pressure response when reproduced by the loudspeakers in the sound system. The desired sound-pressure response creates an apparent sound image location with respect to a listener. The image-corrected signals can also be spatially-enhanced to broaden the apparent sound image and improve the low frequency characteristics of the sound when played on small loudspeakers.

Abstract: A method of encoding stereo audio that minimizes a number of pieces of side information required for parametric-encoding and parametric-decoding of the stereo audio. The side information may include parameters about interchannel intensity difference (IID), interchannel correlation (IC), overall phase difference (OPD), and interchannel phase difference (IPD), which are required to restore the mono audio to the stereo audio.

Abstract: Disclosed is a stereo signal processing apparatus, in particular for a digital BTSC television decoder, comprising a sub-channel signal processing section which comprises an input for inputting an input sub-channel signal, a DBX expanding means and an output for outputting an output sub-channel signal. The particularity of the present invention is that said sub-channel signal processing section further comprises a phase error compensating means for correcting a phase error of said DBX expanding means so that at said output of said sub-channel signal processing section the phase of the output sub-channel signal is essentially constant or zero over a predetermined frequency range.

Abstract: The sound reproduction of a multichannel sound reproduction system with a plurality of speakers which is connected to the output of an FM stereo receiver is controlled by a control signal derived from the reception quality. Preferably, the control signal from the FM stereo receiver for controlling the stereo and mono components is also employed to control the multichannel sound reproduction system. For example, stereo, pseudo-stereo or mono reproduction are provided in the multichannel sound reproduction system in response to the stereo component within the output signal.

Abstract: A stereo sound generation method and apparatus to generate a stereo sound, by using 2-channel headphones, earphones, or speakers, from a multi-channel sound signal reproduced through a variety of media such as a DVD-video, and a DVD-audio. The stereo sound generation method of generating a 2-channel stereo sound from a 5.

Abstract: A computer-implemented method of encoding audio includes accessing a plurality of independent audio source streams, each of which includes a sequence of source frames. Respective source frames of each sequence include respective pluralities of pulse-code modulated audio samples. Each of the plurality of independent audio source streams is separately encoded to generate a plurality of independent encoded streams, each of which corresponds to a respective independent audio source stream. The encoding includes, for respective source frames, converting respective pluralities of pulse-code modulated audio samples to respective pluralities of floating-point frequency samples that are divided into a plurality of frequency bands. An instruction to mix the plurality of independent encoded streams is received; in response, respective floating-point frequency samples of the independent encoded streams are combined. An output bitstream is generated that includes the combined respective floating-point frequency samples.

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

Abstract: A method for processing an audio signal, comprising the steps of extracting an ancillary signal for generating the audio signal and an extension signal included in the ancillary signal from a received bit stream, reading length information for the extension signal, skipping decoding of the extension signal or not using a result of the decoding based on the length information, and generating the audio signal using the ancillary signal. Accordingly, in case of processing the audio signal by the present invention, it is able to reduce a corresponding load of operation to enable efficient processing and enhance a sound quality.

Abstract: Apparatuses, methods, and systems directed to enhancing sound quality in audio signal transmissions. Some embodiments of the present invention comprise an audio signal processor operable to boost one or more frequency components of an input signal. In one embodiment, the audio signal processor boosts three frequency components comprising frequency bands centered near 300 Hz, 1.7 kHz, and 5.4 kHz. Other embodiments of the present invention can be used to enhance stereo audio signals comprising a left and a right input signal. The left and the right input signals are filtered and one or more frequency components of the left and the right input signals are boosted by one or more predetermined values. Some embodiments of the present invention comprise one or more integrated circuits comprising one or more analog or digital filters operable to boost one or more frequency components of an audio input signal.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: A gaming system and related methods comprising a gaming device and a voucher mechanism in communication with the gaming device. The voucher mechanism is configured to present various types of vouchers to the player depending on a game outcome. The types of vouchers include, without limitation, a jackpot voucher, a merchandise voucher, a free play voucher, a mystery voucher, a competition entry voucher, and a restricted machine play voucher.

Abstract: Mid-side (M-S) encoded audio is reproduced by a device that includes a multi-channel digital-to-analog converter (DAC). The DAC has a first channel input receiving a digitized mid audio signal, a first channel output providing an analog mid audio signal, a second channel input receiving a digitized side audio signal and a second channel output providing an analog side audio signal. The DAC may also include a third channel for receiving a digitized second side audio signal. The second side audio signal is phase inverted. The device may be a handheld wireless communication device, such as a cellular phone, and may also include transducers for outputting M-S encoded sound in response to the analog mid and side audio signals.

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: The invention discloses a decoding apparatus for decoding an analog audio signal. The decoding apparatus includes an RF tuner, an analog to digital (A/D) converter, a digital down converter, and a programmable digital signal processor (DSP). The RF tuner is used for receiving the analog audio signal and for providing an analog sound intercarrier frequency (SIF) signal indicative thereof. The analog to digital (A/D) converter is used for sampling the analog SIF signal and for converting the signal into a digital SIF signal. The digital down converter is used for down converting the digital SIF signal to generate a baseband signal. The programmable digital signal processor (DSP) is used for demodulating the baseband signal according to a demodulation procedure of a predetermined standard and decoding the demodulated baseband signal to output an output signal in compliance with a decoding procedure of the predetermined standard.

Abstract: Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information is calculated based on a multi-channel audio signal and a down-mix signal, and a compensation parameter that compensates for the down-mix signal is calculated based on the multi-channel audio signal and the down-mix signal. Thereafter, a bitstream is generated by encoding the spatial information, the compensation parameter, and the down-mix signal and combining the results of the encoding. Therefore, it is possible to prevent deterioration of the quality of sound regarding a multi-channel audio signal by compensating for the multi-channel audio signal using a compensation parameter that compensates for a down-mix signal.

Abstract: A method of processing a signal is disclosed, which includes receiving a downmix signal, object information and rendering information; and generating downmix processing information using the object information and the rendering information, the object information including modified object level information, wherein the modified object level information is determined using a level of a modified object generated by applying a downmix gain to an object. Accordingly, in an apparatus for processing a signal and method thereof according to the present invention, the signal is decoded using object information including modified object level information. Therefore, since a downmix gain is not transmitted, the present invention is able to reduce a used bit for object information and decode a signal using less information.

Abstract: Example embodiments allow for the creation, distribution, and use of flexible media formats. Example embodiments may allow individual content files to be rendered in multiple formats and versions. In addition, example embodiments may provide for granular rights management, which may allow users to access content files on a feature-by-feature basis.

Abstract: An acoustic system that eliminates the howling that occurs when the sound outputted by the speaker feeds back to the input device. The acoustic system comprises a digital signal processor (DSP) that divides the input audio signal into different frequency bands, and reduces the audio levels for the frequency bands where howling is most likely to occur. In one embodiment, the acoustic system comprises a sound source section that generates a test tone that substantially covers the entire human audible range such that the DSP can set the filter levels according to the feedback of the test tone. In another embodiment, the sound source section stores one waveform at a given pitch and generates waveforms of other pitches based on the stored waveform. In yet another embodiment, the pitches of the generated waveforms are dispersed into four frequency bands to create a test tone that resembles a chord or a musical tone.

Abstract: A signal processing apparatus is provided. The signal processing apparatus comprises: an inputting section for inputting audio signals on a plurality of channels; an acoustic type acquiring section which is adapted to acquire an acoustic type of an audio signal on at least one channel of the audio signals; and a process controlling section which is adapted to control a characteristic of sound-field effect applied to the audio signals based on the acquired acoustic type.

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

Abstract: This audio matrix surround decoder requires minimal digital processing, useful in portable applications, particularly in playback from a portable player using a headphone or loudspeaker virtualizer. In one embodiment it pans inputs Lt and Rt to outputs associated with front directions in response to a measure of the sum of Lt and Rt being greater than a measure of the difference between Lt and Rt, and pans Lt and Rt to outputs associated with rear directions in response to a measure of the sum of Lt and Rt being less than a measure of the difference between Lt and Rt. Lt and Rt are modified to shift the direction of reproduced signals.

Abstract: This invention extends standard Bluetooth wireless audio features and capabilities to provide full wireless stereo headset capabilities, and maintain backward compatibility with standard Bluetooth devices. This invention is a full duplex, high fidelity, low latency, two-way digital wireless audio headset with microphone intercom communication system that deploys custom programmed Bluetooth radio transceiver devices.

Abstract: A receiving apparatus comprising: a frequency conversion unit configured to convert a received radio frequency signal to an intermediate frequency signal; an automatic gain control unit configured to control an amplitude level of at least either one of the radio frequency signal and the intermediate frequency signal according to a gain control signal; a demodulation unit configured to demodulate an audio signal from the intermediate frequency signal; a correction level output unit configured to output a correction level signal corresponding to a difference between a demodulation unit input level and a predetermined reference level, the demodulation unit input level being an amplitude level of the intermediate frequency signal inputted to the demodulation unit; and an addition unit configured to add the gain control signal and the correction level signal, to be outputted as a signal strength signal indicating received signal strength of the radio frequency signal.

Abstract: A system and method provide at least a single stage optimization process which maximizes the flatness of the net subwoofer and satellite speaker response in and around a cross-over region. A first stage determines an optimal cross-over frequency by minimizing an objective function in a region around the cross-over frequency. Such objective function measures the variation of the magnitude response in the cross-over region. An optional second stage applies all-pass filtering to reduce incoherent addition of signals from different speakers in the cross-over region. The all-pass filters are preferably included in signal processing for the satellite speakers, and provide a frequency dependent phase adjustment to reduce incoherency between the center and left and right speakers and the subwoofer. The all-pass filters are derived using a recursive adaptive algorithm.

Abstract: A compact audio reproduction system for two input signals includes at least four loudspeakers disposed at the vertices of a quadrilateral not more than two feet on any side and such that no two loudspeakers are located at a distance from one another which is less than one-fourth the greatest distance between any two loudspeakers. The two input signals are connected to alternate speakers such that no two loudspeakers at adjacent vertices of the quadrilateral produce the same signal such that a listener at an arbitrary location perceives a sound source larger than the quadrilateral and significant stereo image. The signals received by two loudspeakers located at adjacent vertices may receive signals which are equalized separately from the signals received by the other loudspeakers for the purpose of reducing comb filtering and improving the tolerance of the device to placement near walls and other obstructions.