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: 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: 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: 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: 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: 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: 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.

Abstract: Certain embodiments of the invention may include systems, methods, and apparatus for controlling sounds in a three dimensional listening environment. According to an example embodiment of the invention, a method is provided for controlling the apparent localization of sounds in a 3-dimensional listening environment. The method can include receiving one or more audio channels, receiving decode data associated with the one or more audio channels, routing the one or more audio channels to a plurality of processing channels, selectively processing audio associated with the plurality of processing channels based at least in part on the received decode data, and outputting processed audio to a plurality of speakers.

Abstract: Electronic devices and accessories such as headsets are provided. An accessory may include speakers and active noise cancellation circuitry. Microphones may be used to pick up ambient noise signals for implementing noise cancellation for the speakers. The accessory may also include a voice microphone and an ambient noise microphone that picks up ambient noise signals for implementing noise cancellation for the voice microphone. A user input interface may gather user input. Ultrasonic tone generators may transmit data between the device and accessory. The electronic device and accessory may be connected to each other by audio connectors. Hybrid circuits that each include a summer and a transconductance amplifier may be selectively switched into or out of use. When switched into use, paths between the device and accessory can support bidirectional communications such as communications involving the simultaneous flow of analog audio and microphone signals in opposite directions.

Abstract: Electronic devices and accessories such as headsets for electronic devices are provided. A microphone may be included in an accessory to capture sound for an associated electronic device. Buttons and other user interfaces may be included in the accessories. An accessory may have an audio plug that connects to a mating audio jack in an electronic device, thereby establishing a wired communications path between the accessory and the electronic device. Path configuration circuitry may be used to selectively configure the path between the electronic device and accessory to support different operational modes. Analog audio lines in the wired path may convey left and right channel analog audio channels. When it is desired to convey power over the wired path, one of the analog audio channel lines may be converted to a power line. Audio functionality may be retained by simultaneously converting a unidirectional line into a bidirectional line using hybrids.

Abstract: There is described a method of encoding input signals (CHI to CH3; 400 to 450) in a multi-channel encoder (5; 15) to generate corresponding output data comprising down-mix output signals (610, 620) together with complementary parametric data (600). The method includes a first step of down-mixing input signals (CHI to CH3; 400 to 450) to generate the corresponding down-mix output signals (610, 620), and a second step of processing the input signals (CHI to CH3; 400 to 450) during down-mixing to generate said parametric data (600) complementary to the down-mix output signals (610, 620). Processing of the input signals (CHI to CH3; 400 to 450) involves including information in the down-mix signals (610, 620) which is useable during subsequent decoding of the down-mix output signals (610, 620) and the parametric data (600) to determine at least some parameter data and thereby enabling representations of the input signals (CHI to CH3; 400 to 450) to be subsequently regenerated.

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: An apparatus and a method of reproducing a wide stereo sound by widening a stereo sound output by an audio reproducing apparatus using only two closely disposed channel speakers include a widening filtering operation and a direct filtering operation. In the widening filtering operation, virtual sound sources for arbitrary locations are formed from a stereo-channel audio signal using head related transfer functions measured at predetermined locations, and crosstalk is cancelled from the virtual sound sources using filter coefficients in which the head related transfer functions are reflected. In the direct filtering operation, signal characteristics of the stereo-channel audio signal are adjusted based on the crosstalk-cancelled virtual sound sources.

Abstract: An audio signal processing circuit for an audio reproduction apparatus at least having sound source located substantially at left and right sides to a listener, is provided. The audio signal processing circuit includes a phase difference control portion. The phase difference control portion receives a left channel signal for the left sound source and a right channel signal for the right sound source, controls a phase difference between the left and right channel signals so as to produce a relative phase difference in the range of 140 degrees to 160 degrees, and outputs the phase difference controlled left and right channel signals for the left and right sound source, respectively.

Abstract: In an audio signal processing device, a signal input part receives a plurality of audio signals to be provided to a plurality of speakers, respectively, arranged so as to surround a listener, the speakers including a center speaker, a left speaker and a right speaker. A signal processing part adds a processed audio signal to an audio signal to be provided to the center speaker, the processed signal being obtained by attenuating a summation of audio signals to be provided to the left speaker and the right speaker. The signal processing part attenuates the summation of the audio signals by an attenuation rate which is set between 0 and 1. The signal processing part sets the attenuation rate to an appropriate value effective to suppress crosstalk between sound emitted from the left speaker and sound emitted from the right speaker.

Abstract: The invention relates to a method and a device for reproducing a binaural output signal generated from a monaural input signal and comprising a first output signal and a second output signal via at least a first and a second speaker of a binaural headset particularly for VoIP applications.

Abstract: The present invention provides an audio apparatus including: a decoding device configured to decode results of channel-by-channel reproduction of a multichannel sound source made up of at least a left channel, a right channel, and a center channel, and to down-mix the decoded results of channel-by-channel reproduction in accordance with the number of speakers configured in a speaker system without a center speaker corresponding to the center channel; an audio processing device configured to perform predetermined audio processing on the decoded results having undergone the down-mixing by the decoding device, and to get the processed results output from the speaker system; and a control device configured to control the audio processing device.

Abstract: A sound effect power supply configuration includes a USB power supply source, an external audio source, a sound effect unit and an external speaker, wherein the USB power input terminal provides sound effect unit USB power source, for the sound effect unit after obtaining USB power source may receive the audio signals output by the external audio source, and after appropriately processing the audio, may drive the external speaker to generate sounds having high quality sound effect.

Abstract: A method and machine-readable medium for providing virtual spatial sound with an audio visual player are disclosed. Input audio is processed into output audio having spatial attributes associated with the spatial sound represented in a room display.

Abstract: The present invention relates generally to sound reproduction systems and methods and more specifically to the reproduction of low-frequency signal components recorded in a Low Frequency Effects (LFE) channel.

Abstract: In the case of binaural coverage, the perception with one-sided sound reception should be improved. Provision is made for this purpose to transmit the receive signals of the hearing devices alternately and if necessary to generate a mono signal from the input signals with a specific weighting, said mono signal being presented at both ears. It is likewise conceivable to use only those spectral components of the input signals to generate the mono signal, said spectral portions having the higher level in each instance.

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 the response of an exponential transfer function.

Abstract: In an embodiment, a phase compensation system shifts the phase of an audio signal in a mid-range frequency band to compensate for phase distortion created when an electrical audio signal is converted to audio by an electronic transducer, such as a loudspeaker. An audio enhancement system mixes at least the phase compensated signal, an enhanced audio signal, and the left and right audio input signals to generated phase compensated left and right audio output signals.

Abstract: The two-way wireless speaker system of this invention increases sound fidelity by enabling speakers to acknowledge receipt of audio data packets. This provides increased functionality because the audio hub can receive data not only from wired inputs, but also wireless transmission from computer, cell phone, and other sources. Audio hub can use information from speaker to customize/adjust audio signal for each speaker independently, giving better audio quality and synchronization among speakers.