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

Abstract: A microphone comprises a movable diaphragm 2 and a back electrode 4. The mechanical relationship between the back electrode 4 and the diaphragm 2 is adjustable thereby to control the cut-off frequency of the microphone. This enables the microphone to be adapted to different noise environments.

Abstract: There is provided a sound processing apparatus and a sound processing method which are capable of reproducing discrete data with a high-quality sound matching users' preferences. In a sound processing means 2, since an interpolation value reflecting a value of a variable parameter ? by which the value of a control sampling function c0(t) is multiplied can be calculated, an analog signal obtained through the interpolation performed in a sampling function sN(t) can be regulated in accordance with the variable parameter ? by changing the value of the variable parameter ?. In this way, by allowing the user to appropriately change the variable parameter ? in accordance with various conditions including music reproduction environments, sound sources, musical tones and so on, it becomes possible to reproduce high-quality-sound music in which its frequency characteristics of the analog signal have changed and a high quality desired by the user is obtained.

Abstract: A speech enhancement system improves the perceptual quality of an aural signal. A receiver detects and receives an unvoiced signal, a fully voiced signal, or a mixed voice remote signal. A coherence processor identifies the similarities or differences between a local signal and the remote signal. A cancellation processor or controller dampens reflected signals that may be part of the local signal.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: An apparatus for performing processing of an input acoustic signal to be reproduced by a loudspeaker, which generates a harmonic of a low pitch sound component equal to or lower than a predetermined low cutoff frequency, and generates a harmonic synthesized acoustic signal synthesizing the input signal with the harmonic. The apparatus generates an output acoustic signal which cuts off, from the harmonic synthesized acoustic signal, a low pitch sound component equal to or lower than the low cutoff frequency and a high pitch sound component equal to or higher than the high cutoff frequency. The apparatus sets a low and high cutoff frequencies in accordance with an output characteristic of a loudspeaker.

Abstract: A method and apparatus to enhance a low frequency component of an audio signal, by computing a fundamental frequency of an input audio signal using the input audio signal and a delayed audio signal obtained by delaying the input audio signal by a predetermined amount of time, generating harmonic signals from the input audio signal based on the fundamental frequency, and combining the harmonic signals and the input audio signal. The low frequency component of the audio signal can be enhanced using human characteristics of perception without physically boosting the energy of the low frequency component.

Abstract: A processing apparatus which is suitable for signal communication with a system having at least one of an input module and an output module. The processing apparatus can be configured to receive input signals from the input module of the system. The processing apparatus includes a first channel processing portion to which the input signals are communicable. The first channel processing portion can be configurable to receive and process the input signals in a manner such that bass frequency audio signals are extracted from the input signals. The bass frequency audio signals can be further processed via at least one of linear dynamic range processing, manipulation of dynamic range via compression and manipulation of dynamic range via expansion.

Abstract: An apparatus for determining a plurality of local center of gravity frequencies of a spectrum of an audio signal includes an offset determiner, a frequency determiner and an iteration controller. The offset determiner determines an offset frequency for each iteration start frequency of a plurality of iteration start frequencies based on the spectrum of the audio signal, wherein a number of discrete sample values of the spectrum is larger than a number of iteration start frequencies. The frequency determiner determines a new plurality of iteration start frequencies by increasing or reducing each iteration start frequency of the plurality of iteration start frequencies by the corresponding determined offset frequency. The iteration controller provides the new plurality of iteration start frequencies to the offset determiner for further iteration or provides the plurality of local center of gravity frequencies, if a predefined termination condition is fulfilled.

Abstract: An audio input signal is processed to produce a processed audio output signal. An audio input signal is received as an original signal. The audio input signal is dynamically filtered to produce a first stage signal consisting of a selected frequency band of the input signal. Gain applied to the first stage signal is dynamically controlled in response to a control signal to produce a second stage signal. The control signal is derived from the first stage signal. Processing the original signal in combination with the second stage signal to produce a processed audio output signal. Processing the original signal in combination with the second stage signal and the first stage signal to produce a processed audio output signal.

Abstract: The present invention relates to an audio equipment (1) and a signal processing method thereof, wherein the mismatches that can occur at the packet boundaries in an audio signal, which is processed in packets by utilizing the missing fundamental phenomenon and which is formed again by the concatenating of these packets, are eliminated.

Abstract: According to one embodiment, an acoustic correction apparatus includes an input module, a calculator, a divider, a converter, an extractor, a synthesizer, and a generator. The input module receives an audio signal propagated through a sound field. The calculator calculates an impulse response from the audio signal. The divider divides the impulse response into first and second impulse responses. The converter converts the first and second impulse responses into first and second frequency spectrums. The extractor specifies an amplitude component of the first frequency spectrum with a peak relatively higher than that of the amplitude component of the first frequency spectrum, and extracts the peak as a resonance component. The synthesizer synthesizes a first property and a second property for attenuating the resonance component. The generator generates a correction filter for performing correction to obtain the synthesized property.

Abstract: An absolute-value circuit outputs the absolute value of a signal SIN? that corresponds to an input signal SIN. A clipping circuit clips the signal SIN? that corresponds to the input signal, to a positive limit value and to a negative limit value. A first multiplier multiplies the signal SIN? that corresponds to the input signal, by a predetermined coefficient. A first adder subtracts the output signal of the first multiplier from the output signal of the clipping circuit. A second adder sums a signal that corresponds to the output signal of the first adder and a signal that corresponds to the output signal of the absolute-value circuit. A third adder sums the input signal SIN and a signal that corresponds to the output signal of the second adder.

Abstract: A frequency characteristics control device of a mixer mixes a first audio signal and a second audio signal inputted to the mixer. In the frequency characteristics control device, a characteristics detection section detects a first frequency characteristic of the first audio signal and a second frequency characteristic of the second audio signal. Based on the first and second frequency characteristics, a removal band detection section detects a removal band in which a level of the first audio signal is higher than a level of the second audio signal. A filtering process section performs a filtering process on the second audio signal so as to attenuate a component of the second audio signal in the removal band. A output section mixes the first audio signal inputted to the mixer and the second audio signal on which the filtering process section has performed the filtering process.

Abstract: A signal conditioner for subwoofers has a band pass filter which passes a lower frequency spectrum of an input signal. The band pass filter includes a high pass filter having a predetermined corner frequency, and a low pass filter having about the same corner frequency, and produces a filtered input to a peak voltage follower circuit. The peak voltage follower circuit traces and smoothes the upper boundary of the filtered input to produce a control voltage. An analog multiplier multiplies the input signal by the control voltage produced by the peak voltage follower to produce an enhanced signal where low frequency transients have been magnified.

Abstract: An audio signal mixer can be applied to a DJ device and the like, for example. The audio signal mixer acquires a plurality of audio signals and acquires mixing rate information being information on a mixing rate of a plurality of the audio signals. The audio signal mixer decides an adjusting amount of balance between levels at every band when a plurality of the audio signals are mixed based on the mixing rate. A plurality of the audio signals are mixed at every band, adjusts the levels of the mixed signals of the respective bands based on the adjusting amount, adds the adjusted signals of the respective bands and outputs it as a mixed audio signal.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: A method for adjusting frequency response curve of a speaker comprises these steps: testing sensitivity of the speaker in at least two types of hardware conditions and recording corresponding frequency response curves; selecting a frequency response curve that comes closest to falling within a predetermined range for selected frequency ranges; and adjusting the selected frequency response curve with a filter.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: A delivery system is disclosed, which has a transmission apparatus and a plurality of reception terminals communicably connected through a network. The transmission apparatus includes a correction section which pre-obtains a plurality of types of correction coefficients according to acoustic characteristics of the plurality of reception terminals and forms a plurality of corrected audio signals based on an audio signal and the correction coefficients; a conversion section which converts the plurality of corrected audio signals into a plurality of pieces of delivery data; and a delivery server which delivers the delivery data to the reception terminals through the network. Each of the reception terminals includes a communication section which communicates with the transmission apparatus. The transmission apparatus delivers to the reception terminals the delivery data according to the acoustic characteristics thereof.

Abstract: A combined multirate-based Finite Impulse Response (FIR) filter equalization technique combines a low-order FIR equalization filter operating at a lower rate for equalization of a loudspeaker-room response at low frequencies, and a complementary low-order minimum-phase FIR equalization filter operating at a higher rate for equalization of the loudspeaker-room response at higher frequencies. The design of two complementary band filters for separately performing low and high frequency equalization, keeps the system delay at a minimum while maintaining excellent equalization performance. Splicing between the two equalization filters, for maintaining a flat magnitude response in the transition region of the two complementary filters, is done automatically through level adjustment of one equalization filter relative to the other. The present invention achieves excellent equalization at low filter orders and hence reduced computational complexity.

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

Abstract: This invention relates to effects processing of a monophonic analog signal, meaning a signal whose frequency components are all integer multiples of a first fundamental frequency. For example, the signal could come from almost any musical instrument, voice included. However, for generality, the invention is not restricted to cases where the signal source is musical. The digital signal processing is simplified as a result of the DSP being clocked at a constant multiple of ffund. This means that the sine and cosine functions, as well as the low-pass filters which make up each harmonic selector, are trivial to implement because the frequencies of each sine/cosine, as well as the cutoff frequency of the low-pass filters, are constant fractions of the DSP clock frequency.

Abstract: An audio test apparatus, and an exemplary audio test method that includes: processing an audio file through two independent channels; outputting no signals from a left channel and from a right channel in a first time period; receiving noise signals from the left and right channels; outputting single-frequency signals from the left channel only in a second time period; receiving the single-frequency signals from the left channel and crosstalk signals from the right channel; outputting multi-frequency signals from the left and right channels in a third time period; receiving the multi-frequency signals from the left and right channels; outputting single-frequency signals from the right channel only in a fourth time period; receiving the crosstalk signals from the left channel and the single-frequency signals from the right channel; and testing parameters according to the signals received during the four time periods.

Abstract: A system and method of producing a directional output signal is described including the steps of: detecting sounds at the left and rights sides of a person's head to produce left and right signals; determining the similarity of the signals; modifying the signals based on their similarity; and combining the modified left and right signals to produce an output signal.

Abstract: An apparatus is provided for controlling a graphic equalizer which is implemented by digital filters, each being assignable to a frequency band for regulating a level of the frequency band. In the apparatus, a set of control devices are provided in correspondence to respective frequency bands, each control device being operable to specify the level of the corresponding frequency band to either of a reference level or other level than the reference level. An assignment section operates when a number of the digital filters is less than a total number of the frequency bands, for assigning the digital filters to the frequency bands which are specified with the levels other than the reference level. A control section controls remaining digital filters which are not assigned to any of the frequency bands in a through state.

Abstract: A method of compensating audio frequency response characteristics of a portable sound system can be performed using acoustic characteristics of a user measured in real-time. The method of compensating the audio frequency response characteristics can be used by a portable sound system. The method includes generating an acoustic characteristics curve of a user based on a minimum perception level of a user with respect to audible audio frequency bandwidths, generating an acoustic compensation curve of the user based on the acoustic characteristics curve of the user and a predetermined frequency characteristics target curve, and compensating the audio frequency response characteristics of a sound based on the acoustic compensation curve of the user.

Abstract: According to one embodiment, an acoustic characteristic correction coefficient calculation apparatus includes a frequency converter, a smoother, a frequency inverter, a cutter, and a calculator. The frequency converter is configured to convert a first impulse response corresponding to an input acoustic signal to a frequency domain. The smoother is configured to smooth amplitude and phase corresponding to the frequency domain converted by the frequency converter. The frequency inverter is configured to convert a frequency characteristic smoothed by the smoother to a time domain. The cutter is configured to cut out a second impulse response configured by the time domain obtained by converting the frequency characteristic by use of the frequency inverter by a preset tap number. The calculator is configured to calculate a correction coefficient used to correct an acoustic characteristic based on a result cut out by the cutter.

Abstract: A system for boosting the bass of an audio signal to closely match or mirrors a plurality of Robinson-Dadson loudness curves by interpolating coefficients from a table of values representing the Robinson-Dadson loudness. The system having a controller that interpolates the coefficients from the loudness curves and then uses the coefficients in a shelf filter that makes adjustments to the audio signal. The result of the adjustments to the audio signal is the introduction of bass boost slowly through a diminuendo or lowering of level through volume adjustment and to removes the bass boost rapidly during a crescendo or increase in level through user volume adjustment.

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: A system and method for monitoring for a specified frequency band is disclosed. The technology initially utilizes a micro-electromechanical system (“MEMS”) based acquisition device to monitor an environment. The MEMS device receives a signal from the environment, and generates an input signal comprising an electronic representation of the received environmental signal. This input signal is then conditioned for at least one frequency band. Embodiments of the invention next allow the conditioned signal to be compared to various pre-defined events in order to determine the signal's origin.

Abstract: An audio signal supplying device includes a connection terminal which is connected to a speaker unit and which outputs an audio signal to the speaker unit, a specifying section which specifies a type of the speaker unit connected to the connection terminal, an acquiring section which acquires a parameter corresponding to the type of the speaker unit specified by the specifying section from an outside, a storing section which stores the parameter acquired by the acquiring section, an inputting section into which the audio signal is input, an acoustic characteristic controlling section which adjusts an acoustic characteristic of the audio signal input into the inputting section based on the parameter stored in the storing section, and a supplying section which supplies the audio signal whose acoustic characteristic is adjusted by the acoustic characteristic controlling section to the connection terminal.

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: Provided are a method and apparatus for controlling a sound field through an array speaker. The method includes calculating a coefficient of a filter that controls sound pressure of an input signal, based on a sound pressure ratio of a suppression area that suppresses sound emitted from an array speaker and an emphasis area that emphasizes the sound, and sound pressure efficiency in the emphasis area, generating a plurality of output signals that focuses the sound to the emphasis area by filtering the input signal based on the calculated coefficient of the filter, and outputting a sound field controlled sound based on the generated plurality of output signals. Accordingly, a listener in a predetermined direction and distance from the array speaker can clearly hear sound, without wearing an earphone or a headset so as to focus the sound only to the listener.

Abstract: A method (200) for improving quality of output audio (126). The method can include detecting an output acoustic signal (128) and generating a receive audio signal (134) based, at least in part, on the detected output acoustic signal. A frequency domain representation (140) of the receive audio signal can be compared to a frequency domain representation (138) of a source audio signal (124) from which the output acoustic signal is generated. At least one distortion signal (142) in the receive audio signal can be identified, and the source audio signal can be selectively equalized to reduce an amplitude of the source audio signal at a frequency that correlates to the distortion signal.

Abstract: LPF and HPF extract bass and treble ranges, respectively, from an input sound signal, and bass and treble boost circuits perform dynamic range expansion/contraction on the extracted bass- and treble-range sound signals in accordance with input levels of the sound signals. The input sound signal and the sound signals output from the boost circuits are added together. There may also be provided coefficient calculation sections for calculating filter coefficients on the basis of the levels of the sound signals extracted by the LPF and HPF. In this case, the bass and treble boost sections perform, in accordance with the filter coefficients calculated by the corresponding coefficient calculation sections, filter processes for increasing/decreasing the levels of the bass and treble ranges, respectively.

Abstract: According to one embodiment, a signal correction apparatus includes a sound quality estimation module, a sound quality enhancement processing controller, and a sound quality enhancement processor. The sound quality estimation module is configured to generate channel importance information based on correlations of channels for signals input from one or more channels, and to generate frequency importance information based on noise levels. The sound quality enhancement processing controller is configured to decide valid channels and valid frequency components which are to undergo sound quality enhancement processing of the signals based on the channel importance information and the frequency importance information, and to generate valid channel information and valid processing frequency information.

Abstract: The application relates to a method of improving a user's perception of an input sound. The application further relates to an audio processing device and to its use. The object of the present application is to increase the sound quality of a sound signal as perceived by a user, e.g. a hearing impaired user. The method comprises a) Defining a critical frequency fcrit between a low frequency range and a high frequency range; b) Analyzing an input sound in a number of frequency bands below and above said critical frequency; c) Defining a cut-off frequency fcut below said critical frequency fcrit; d) Identifying a source frequency band above said cut-off frequency fcut; e) Extracting the envelope of said source band; f) Identifying a corresponding target band below said critical frequency fcrit; g) Extracting the phase of said target band; h) Combining the envelope of said source band with the phase of said target band.

Abstract: An apparatus and method of canceling a vocal component includes a first vocal canceling unit to delay each of the left and right channel input signals for a predetermined time and to feed-forward cross mix the delayed left and right channel signals with the left and right channel input signals, a sound stage widening unit to delay each of the left and right channel signals output from the first vocal canceling unit for a predetermined time and to feedback cross mix the signal of each delayed channel signals with the left and right channel signals, and a second vocal canceling unit to low-pass filter the left and right channel signals output from the sound stage widening unit and to mix the low-pass filtered left and right low frequency components and a high frequency component of the difference between the left and right channels to cancel the vocal component from an audio signal.

Abstract: A computer program product for computing a correction rate predictor for medical record dictations, the computer program product residing on a computer-readable medium includes computer-readable instructions for causing a computer to obtain a draft medical transcription of at least a portion of a dictation, the dictation being from medical personnel and concerning a patient, determine features of the dictation to produce a feature set comprising a combination of features of the dictation, the features being relevant to a quantity of transcription errors in the transcription, analyze the feature set to compute a predicted correction rate associated with the dictation and use the predicted correction rate to determine whether to provide at least a portion of the transcription to a transcriptionist.

Abstract: A method of setting an equalizer so as to enlarge a sound field in reproducing an audio file and a method of reproducing an audio file thereby, includes: dividing an input audio file into segments with a predetermined time length; extracting an audio feature value for each segment; determining equalizer information for reproducing each segment by the use of the extracted feature value; and determining an equalizer sequence for the audio file by the use of the determined equalizer information of each segment. Since the equalizer setting information can be automatically changed without user manipulation, the user can listen to an audio file of which the sound field is dynamically enlarged.

Abstract: Described herein are systems and methods for processing audio data. For example, one embodiment provides a device configured to receive an audio signal, and process that audio signal based on a frequency mapping protocol, thereby to provide a modified audio signal. The resulting modified signal is audibly communicated to a user. In overview, the frequency mapping protocol allows components of an audio signal that within specified frequency ranges to be mapped to other frequency ranges, attenuated, or in some cases blocked altogether. The user is also able, at least in some embodiments, to use the same device to perform testing of their hearing. In some embodiments the frequency mapping protocol is modifiable by a user, for example by way of a software application running on a PC or other computational platform. This allows a significant degree of flexibility in terms of the device's operation; a user is effectively able to customize the device for his/her specific hearing requirements.

Abstract: A method and system is described to improve the reproduction of sound of an acoustic musical instrument. According to one embodiment, a first microphone is placed at a proximate location to the musical instrument to pick up the sound of the musical instrument. The sound as picked up by the first microphone is compared to a reference sound of the instrument (e.g., the sound of the instrument as perceived at a normal listening position). Based on this comparison, a tailor-made equalizer is designed to compensate for the differences between the sounds as picked up by the first microphone and the reference sounds of the musical instrument. Accordingly, using the tailor-made equalizer allows the reproduction of sound from the first microphone to have a quality similar to that of the reference sound of the musical instrument. In an implementation of the above system, a filter arrangement is provided having a low-pass and a high-pass filter that allows separate control of the frequency and/or gain for each filter.

Abstract: An apparatus for generating a bandwidth extended signal from an input signal includes a patch generator and a combiner. The input signal is represented for first and second bands by first and second resolution data, respectively, the second resolution being lower than the first. The patch generator generates first and second patches from the first band of the input signal according to first and second patching algorithms, respectively. A spectral density of the second patch generated using the second patching algorithm is higher than a spectral density of a first patch generated using the first patching algorithm. The combiner combines both patches and the first band of the input signal to obtain the bandwidth extended signal. The apparatus scales the input signal according to the first and second patching algorithms or scales the first and second patches, so that the bandwidth extended signal fulfills a spectral envelope criterion.

Abstract: A vehicle engine harmonic enhancement system. A method for operating the system includes providing a first engine harmonic enhancement audio signal and equalizing the first engine harmonic enhancement audio signal separately for each of the plurality of loudspeakers to provide individually equalized loudspeaker engine harmonic enhancement audio signals. A vehicle audio system includes a plurality of loudspeakers, an entertainment audio system, and an engine harmonic enhancement system. A method includes equalizing an entertainment audio signal to provide an equalized entertainment audio signal; providing an engine harmonic enhancement audio signal; equalizing the engine harmonic enhancement audio signal separately from the entertainment audio signal to provide an equalized engine harmonic enhancement audio signal; and combining the equalized entertainment audio signal with the equalized engine harmonic enhancement audio signal.

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 speaker array, includes a plurality of speakers which are linearly arranged at a predetermined interval; and one-dimensional digital filters which are provided to correspond to the speakers respectively, in which predetermined filter coefficients are set previously, and which apply a filtering process to input sound data in response to the filter coefficients to output. Sound data derived by applying a digital conversion to input sound signals are supplied to respective one-dimensional digital filters. Sound signals derived by applying an analog conversion to the sound data output from respective one-dimensional digital filters are supplied to corresponding speakers to output a sound in response to the sound signals.

Abstract: A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user.

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.