Abstract: A sound signal processing device causes an amplifier to amplify a low frequency signal and synthesizes the amplified low frequency signal with a middle or high frequency signal so as to output the resultant output signal. A gain control section controls a gain of the amplifier on the basis of an amplitude of the output signal. As a result, it is possible to amplify the low frequency signal without any clipping in a receiving end device and without any unstable volume of a middle or high frequency component.

Abstract: A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

Abstract: The interference powers with directional microphones are to be suppressed as far as possible. To this end, provision is made to adaptively filter the microphone of a number of microphones as a function of at least one parameter. The directional effect of the directional microphone achieved in this way is adjusted by modifying the at least one parameter, such that the summation of interference powers including microphone noises is reduced and/or minimal.

Abstract: A method for performance measurement and optimization of sound systems using electroacoustic measurements and a sliding band integration curve. Nearfield and spatially and temporally averaged broadband farfield responses are measured, averaged over a distinct set of frequencies, level matched, and weighted using a frequency-dependent ratio. The two curves are then combined to produce a third curve. The results indicate system performance in a listening space that matches human sensory response and provides means to optimize the sound system for the listening space.

Abstract: A device and method for improving the performance of an electro-acoustic transducer. An acoustic test signal is generated through the electro-acoustic transducer. The acoustic test signal is measured at multiple points on an ambient surface around the electro-acoustic transducer to create measured acoustic data. Based on the measured acoustic data, an acoustic power frequency response of the electro-acoustic transducer is calculated. A correction impulse response for the electro-acoustic transducer is determined based on the acoustic power frequency response. A correction filter applies the correction impulse response to a sound signal input to generate a sound signal output for playback through the electro-acoustic transducer.

Abstract: A speaker device comprises a first speaker (SP1) for reproducing an audio signal and a second speaker (SP2) adapted for reproducing an audio signal and spaced from the first speaker horizontally by a predetermined distance. At least one of the first and second speakers phase has varying means (APF1, APF2, . . . ) for varying the phase of the audio signal by a predetermined quantity of phase according to (i) the frequency of the audio signal and (ii) the predetermined distance.

Abstract: The sound reproducing apparatus for use with a signal source comprises, a first output unit connected to the signal source via a first power amplifier, a first phase shifter connected to the signal source, a second output unit connected to the first phase shifter via a second power amplifier, a second phase shifter connected to the signal source, a third output unit connected to the second phase shifter via a third power amplifier, a fourth phase shifter connected to a third phase shifter, and a fourth output unit connected to the fourth phase shifter via a fourth power amplifier, wherein the first output unit and the second output unit are in right and left relation with each other, the third output unit and the fourth output unit are in right and left relation with each other, a group of the first output unit and the second output unit and a group of the third output unit and the fourth output unit are in front and rear relation with each other. The vehicle is equipped with the sound reproducing apparatus.

Abstract: An audio phase corrector circuit which shifts the phase of substantially all frequencies of an audio signal by 90 degrees for the purpose of enhancing sound produced by an electroacoustic transducer. Embodiments are implemented by a digital finite impulse response filter configured to operate as a 90-degree phase shift circuit, commonly called a Hilbert transformer. The 90-degree phase shift corrects a characteristic phase distortion caused by electroacoustic transducers and thereby retains much of the impulse and transient information in the acoustic output that would otherwise be lost. The resulting sound has substantially improved clarity, detail, presence, placement, and spaciousness.

Abstract: A method for synchronizing audio reproduction in collocated end devices is presented. Each of the devices auto-correlates using noise to determine a threshold prior to the antenna receiving an audio signal. When the devices receive a common audio signal, they provide audio outputs. Each device cross-correlates its audio output with the audio outputs of the other devices. The timing of the audio output of each device is then adjusted such that the audio outputs of all of the devices align temporally with the lagging or leading device.

Abstract: A sound processing apparatus (100), which can improve precision of analyses on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors (first sound collector 110-1 and second sound collector 110-2), and the sound processing apparatus (100) is provided with a level signal conversion section (first level signal conversion section 130-1, second level signal conversion section 130-2) that converts the collected sound signal into a level signal, from which phase information is removed, a level signal synthesizing section (140) that generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors (first sound collector 110-1 and second sound collector 110-2) are synthesized, and a detecting and identifying section (160) that carries out analysis on the ambient sounds based upon the synthesized level signal.

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

Abstract: A sound signal processing apparatus and a sound signal processing method divide an input signal into a low frequency signal output (11) and a high frequency signal output (10), and delay only the high frequency signal output (10), thereby reducing the temporal shift between the high frequency signal and the low frequency signal. Furthermore, correcting the phase of the low frequency signal output (11) in accordance with a change in phase due to the delay of the high frequency signal output (10) allows to prevent variation in frequency characteristics due to interference at the time of addition of the low frequency signal output (11) and the high frequency signal output (10).

Abstract: An audio apparatus includes an input, a first resistor, a first capacitor, an amplifier, a second resistor, a second capacitor, and an output. The input is used for inputting audio signals. The first resistor and the amplifier are serially connected to the input; wherein the first resistor is connected to the inverting input of the amplifier and the non-inverting input of the amplifier is connected to ground. The first capacitor has one end connected to a node between the first resistor and the inverting input of the amplifier, the other end connected to ground. The second resistor has one end connected to a node between the first resistor and the inverting input of the amplifier, the other end connected to the output of the amplifier. The second capacitor is connected between the inverting input and the output of the amplifier. The output is connected to the output of the amplifier, for outputting the audio signals after processing.

Abstract: An audio processing apparatus including an audio phase detecting device and an adjusting device is provided. After detecting a phase relationship between a first channel signal and a second channel signal, the audio phase detecting device generates a phase control signal. The adjusting device is coupled to the audio phase detecting device and used for selectively adjusting the first channel signal according to the phase control signal.

Abstract: An audio signal delivery management system includes a management apparatus and an delivering apparatus which delivers an audio signal to a plurality of speaker devices to which respective addresses are assigned. The delivering apparatus include an address storing portion which stores the addresses of the speaker devices, an audio signal acquiring portion which acquires an audio signal, an audio signal delivering portion which delivers the audio signal to at least one of the speaker devices using the addresses, and a sending portion which sends the audio signal to the management apparatus.

Abstract: A device for processing an audio signal is provided, wherein the processing device comprises a processing unit, and a determination unit, wherein the processing unit is adapted to receive first data associated with a first subsignal of the audio signal and second data associated with a second subsignal of the audio signal, and wherein the determination unit is adapted to determine a compensation value for one of the subsignals depending on a phase difference between the first subsignal and the second subsignal.

Abstract: A method and apparatus for increasing phase margin in a feedback circuit of an active noise reduction headphone. The method includes providing an acoustic block comprising an acoustic driver comprising a voice coil mechanically coupled along an attachment line to an acoustic energy radiating diaphragm, the acoustic block further comprising a microphone positioned along a line parallel to an intended direction of vibration of the acoustic diaphragm and intersecting the attachment line, the acoustic block characterized by a magnitude frequency response compensating the magnitude frequency response by a compensation pattern that has a positive slope over at least one spectral range above 10 kHz.

Abstract: Speaker system is constructed of a plurality of speakers each including a bass-reflex type cabinet. The speaker system is designed to differentiate a low-band resonance frequency between the speakers by differentiating inner cubic capacities of the speaker cabinets from each other. Audio signal of a different channel is input to each of the speakers, and only low-frequency signals of all of the channels are added together so that the added result is supplied to all of the speakers. Thus, using the differentiated low-band resonance frequencies, the speaker system permits reproduction of low-pitched sounds with a flat characteristic.

Abstract: A speaker system that includes an audio signal-receiving interface, a modulating circuit, a phase-control circuit, and a number of speaker units. The audio signal-receiving interface is configured to receive an audio signal, and the modulating circuit is coupled with the audio signal-receiving interface. The modulating circuit is configured to modulate a low frequency component of the audio signal and to generate a modulated signal. The phase-control circuit is coupled with the modulating circuit and the audio signal-receiving interface. The phase-control circuit is configured to receive the modulated signal and a high-frequency component of the audio signal and to control a phase of the modulated signal, a phase of the high-frequency component of the audio signal, or both. The speaker units are coupled with the phase-control circuit and configured to generate sound waves based on signals supplied by the phase-control circuit.

Abstract: A method and apparatus for conditioning an audio input signal to enhance perception and reproduction of bass frequencies. Harmonics are generated and combined with a phase-shifted version of the audio input signal. Use of a controlled phase shift reduces or eliminates unwanted introduction of waveform asymmetry or D.C. offset.

Abstract: A set of filters is configured to distribute input signals representing a single perceptual axis to first and second physically separate arrays of loudspeakers comprising at least first and second transducers, such that the arrays of loudspeakers will create an array pattern corresponding to the input signals when the input signals are between a first frequency and a second frequency.

Abstract: Methods and apparatuses for media playback system are disclosed, including a signal source for providing electrical signal representing media content; a processing unit for processing signal from the signal source; a distorting unit for introducing distortion to signal from the processing unit; a transducer for converting the electrical signal from the distorting unit to energy representing the media content; and a connection link for transmitting the signal from the distorting unit to the transducer. The arrangement protects media content against unauthorized duplication by ripping-off the transmitting signal.

Abstract: Embodiments of the present invention include methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they are can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio objects class is used to give priority to or apply specific processing to audio objects sin the presence of other audio objects of different classes.

Abstract: A mixed audio separation system (100) which separates a specific audio from among a mixed audio (S100) includes a local frequency information generation unit (105) which obtains pieces of local frequency information (S103) corresponding to local reference waveforms (S102), based on the local reference waveforms (S102) and an analysis waveform which is the waveform of the mixed audio (S100). Each of the local reference waveforms (S102) (i) constitutes a part of a reference waveform for analyzing a predetermined frequency, (ii) has a predetermined temporal/spatial resolution and (iii) includes at least one of an amplification spectrum and a phase spectrum in the predetermined frequency.

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: The phase differences of microphones of a hearing aid microphone are to be reduced. To do this, the level of an output signal of a directional microphone is compared with an omnidirectional signal. If the level of the output signal of the differential directional microphone is above the level of the omnidirectional signal, this level difference is minimized by an adaptive, frequency-selective transit time compensation in individual frequency bands and phase matching of the microphones is thus achieved. By means of an alternative method, microphone matching is achieved in that the measurable delay of the two microphone signals is adaptively limited in individual frequency bands to a maximum value corresponding to the sound transit time between the microphones. Phase matching without knowing the position of a sound source can thus be achieved.

Abstract: A speaker arrangement is described, suitable for improving the sound quality of a small size first speaker. The arrangement comprises a cavity, to which the first speaker is acoustically coupled. A second speaker is also acoustically coupled to the cavity, and the first and second speakers are excited in phase with each other. The first speaker may be an earphone speaker tuned to be held to a user's ear, whereas the second speaker may be a larger ring alert speaker with lower compliance than the first speaker, both of which are incorporated in the same mobile phone. By exciting the second speaker in phase with the first speaker, the second speaker will aid the smaller first speaker's reproduction of sound, particularly in the lower frequency range.

Abstract: Provided is a single sound transmission apparatus using human body communication. The single sound transmission apparatus includes a sound signal generator generating a sound signal of an audio frequency band, a high frequency signal generator generating a high frequency signal having a higher frequency than the audio frequency band, a signal combiner generating a combined signal by combining the sound signal with the high frequency signal, and a signal transmitter varying a phase difference between the combined signal outputted from the signal combiner and the high frequency signal outputted from the high frequency signal generator, and outputting the combined signal and the high frequency signal together to a human body, in order to leave only the sound signal through destructive interference of the high frequency signal of the combined signal, wherein the destructive interference is generated in an ear area of the human body.

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: 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: Power amplifies are provided to correspond with a plurality of speaker units constituting a speaker array. Driving signals from the power amplifies are supplied to one terminals of the speaker units, and other terminals are connected a common line. Inverters are provided at preceding stages of the power amplifies such that the driving signals supplied to the speaker units being arranged adjacently have an opposite phase respectively. Since a correlation between the driving signals of the adjacent speaker units is high, a current flowing through the common line can be very reduced. An LED display panel can be driven by the similar method.

Abstract: An acoustic processing device includes an input section to which audio signals of a plurality of channels respectively including in-phase components are input, a phase adjusting section that adjusts phases of the audio signals of the plurality of channels respectively to generate phase adjustment signals of the plurality of channels being different in phase from the audio signals of the plurality of channels input to the input section, an anti-phase generating section that generates an anti-phase signal by adding the phase adjustment signals of the plurality of channels to each other and adjusting a phase of the added signal to a substantially inverted phase, and an output section that outputs signals obtained by adding, to each of the audio signals of the plurality of channels input to the input section, the phase adjustment signal of another channel and the anti-phase signal.

Abstract: An audio reproduction device reproduces a one-channel signal from two speakers arranged in a sound field. The one-channel input signal is delayed by a delay element. The delay element performs a delay with different delay amount for a plurality of frequency bands. The signal delayed by the delay element is outputted to one of the two speakers and the input signal is directly outputted to the other one of the two speakers.

Abstract: An acoustic device including a first acoustic waveguide having two open ends; a second acoustic waveguide; and an acoustic driver having a first and second radiating surface positioned so that the first: radiating surface radiates into the first waveguide and the second surface radiates into the second waveguide. An acoustic device including an acoustic driver and an acoustic waveguide with two open ends. A method for making the acoustic device.

Abstract: A digital microphone interface includes a phase generator for generating a phase signal. A clock generator produces a digital microphone clock at a selected frequency based on the frequency select signal, and supplying the digital microphone clock to a digital microphone. A first data latch, operably coupled to the phase generator, produces a stream of first data from a first channel of the digital microphone, based on a phase select signal and the phase signal.

Abstract: This invention relates to a system (10) and method for matching one or more microphones. A first and second microphone (110, 210) communicates, respectively, a first and second microphone signal to an amplitude compensating means (16) adjusting amplitude of the first microphone signal in accordance with amplitude of the second microphone signal. The amplitude compensating means (16) communicates an adjusted first microphone signal (100) and the second microphone signal (200) to a phase matching means (18) comprising a correction filter means (113) receiving the adjusted first microphone signal (100). The correction filter means (113) comprises a low-pass filter means (111), which is controlled in accordance with a subtraction between the first microphone signal output (102) and the second microphone signal (200), and a high-pass filter means (112), which is controlled in accordance with a comparison between the first microphone signal output (102) and the second microphone signal (200).

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: There is provided a voice recognition device and a voice recognition method that enhance the function of noise adaptation processing in voice recognition processing and reduce the capacity of a memory being used. Acoustic models are subjected to clustering processing to calculate the centroid of each cluster and the differential vector between the centroid and each model, model composition between each kind of assumed noise model and the calculated centroid is carried out, and the centroid of each composition model and the differential vector are stored in a memory. In the actual recognition processing, the centroid optimal to the environment estimated by the utterance environmental estimation is extracted from the memory, model restoration is carried out on the extracted centroid by using the differential vector stored in the memory, and noise adaptation processing is executed on the basis of the restored model.

Abstract: A speaker device comprises a first speaker (SP1) for reproducing an audio signal and a second speaker (SP2) adapted for reproducing an audio signal and spaced from the first speaker horizontally by a predetermined distance. At least one of the first and second speakers phase has varying means (APF1, APF2, . . . ) for varying the phase of the audio signal by a predetermined quantity of phase according to (i) the frequency of the audio signal and (ii) the predetermined distance.

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: A system and method for minimizing the complex phase interaction between non-coincident subwoofer and satellite speakers for improved magnitude response control in a cross-over region. An all-pass filter is cascaded with bass-management filters in at least one filter channel, and preferably all-pass filters are cascaded in each satellite speaker channel. Pole angles and magnitudes for the all-pass filters are recursively calculated to minimize phase incoherence. A step of selecting an optimal cross-over frequency may be performed in conjunction with the all-pass filtering, and is preferably used to select an optimal cross-over frequency prior to determining all-pass filter coefficients.

Abstract: A multi-way loudspeaker system comprising of frequency equalization circuit, a linear phase dividing network, with optional phase shifters to enable the loudspeaker to have substantially improved frequency and phase response characteristics but not limited thereto is discussed. The loudspeaker exhibits high accuracy not only in amplitude but also in time domain characteristics which results in an improved transmission characteristics of time sensitive signals such as square waves.

Abstract: A filter circuit includes: an input terminal; a first resistance; a second resistance; a capacitor; and an output terminal, in which the first resistance, the second resistance, and the capacitor are connected in series in this order between the input terminal and a ground point, the output terminal is provided at a connection point of the first resistance and the second resistance, and a frequency domain is used that is higher than a maximum phase delay frequency higher than a cutoff frequency, the cutoff frequency being determined by a combined resistance value of the first and the second resistances and a capacitance value of the capacitor, so that when a frequency of an input signal becomes higher, a phase delay of an output signal relative to the input signal is reduced.

Abstract: Provide is a forced acoustic dipole in which a direction of an acoustic lobe can be freely steered by controlling phase and sound pressure of input signals input into first and second sound sources, respectively. Accordingly, when a forced acoustic multipole array is configured by arranging a plurality of forced acoustic dipoles and a direction of an acoustic lobe is steered in a particular direction by controlling phase and sound pressure of input signals input into the first and second sound sources, respectively, it is possible for a sound to be heard in a desired location without auditory disturbance to others.

Abstract: A system and method for minimizing the complex phase interaction between non-coincident subwoofer and satellite speakers for improved magnitude response control in a cross-over region. An all-pass filter is cascaded with bass-management filters in at least one filter channel, and preferably all-pass filters are cascaded in each satellite speaker channel. Pole angles and magnitudes for the all-pass filters are recursively calculated to minimize phase incoherence. A step of selecting an optimal cross-over frequency may be performed in conjunction with the all-pass filtering, and is preferably used to select an optimal cross-over frequency prior to determining all-pass filter coefficients.

Abstract: Provided are a human body sound transmission system and a method using single sound source. The human body sound transmission system includes a first transmission block for transmitting a combined signal of a first high frequency signal and a sound signal through a human body; and a second transmission block for transmitting a second high frequency signal having the same frequency as the first high frequency signal through the human body, to thereby recover the sound signal by destructing interference in a ear region of a user.

Abstract: An arrangement includes an input, a modulation circuit, an isolation circuit, a demodulator circuit, a detector circuit and at least two outputs. The input is configurable to receive an audio signal and a DC voltage. The modulation circuit is configured to modulate the audio signal and the DC voltage. The isolation circuit has an input electrically isolated from the output, wherein its input coupled to the modulation circuit. The demodulator circuit is operably coupled to the output of the isolation circuit and configured to extract the audio signal therefrom. The detector circuit is operably coupled to the output of the demodulator, and is configured to generate a first signal responsive to detection of a modulated signal corresponding to the DC voltage modulated by the modulation circuit.

Abstract: An acoustic signal processing apparatus includes an acoustic signal input device, a frequency resolution device, a two-dimensional data generating device, a graphics detection device, a sound source candidate information generating device, and a sound source information generating device. The sound source information generating device generates sound source information including at least one of the number of sound sources, the spatial existence range of the sound source, an existence period of the voice, a frequency component configuration of the voice, amplitude information on the voice, and symbolic contents of the voice based on the sound source candidate information and corresponding information which are generated by the sound source candidate information generating device.

Abstract: The invention relates to an automated system for adjusting line array speakers. The automated system includes a system for moving two or more speakers with a moving device. Additionally, moving two linear actuators essentially simultaneously; a bracket to attach to the moving device to the speaker; a remote control system for controlling the movement of the speakers and for displaying the position of the speakers in real time; and a system for modeling and determining the proper frequency response for a venue and automatically adjusting the linear array speaker systems to the proper position for the proper frequency response.

Abstract: In a sound signal playback machine and method thereof, predetermined high frequency components are extracted from sound signals SL to SC of a main channel by high frequency pass filters 21 to 25, and these high frequency components are respectively played back by speakers. At the same time, predetermined low frequency components are extracted from sound signals SL to SC of the main channel by low frequency pass filters. These low frequency components and sound signal SLFE exclusively used for the low frequency band are added to each other, and the thus obtained addition signal is played back by the speaker 66. In this case, the degree of the low frequency pass filters is set higher than the degree of the high frequency pass filters, and the high frequency component is delayed.