Abstract: A method for enhancing the accuracy of sounds heard by a hearing-impaired listener is disclosed. The method for enhancing the accuracy of sounds heard by a hearing-impaired listener includes receiving an input sound, determining if it is necessary to modify the frequency of the input sound, and modifying the input sound into a modified input sound if necessary. The determination relies on the frequency and energy of the input sound. The ratio of the energy of lower frequencies of the modified input sound will be increased.

Abstract: A loudness-compensating volume control method imposes a desired loudness scaling on an audio signal by processing the audio signal in both the digital and analog domains by receiving a desired loudness scaling, deriving a wideband gain component and one or more other gain components from the desired loudness scaling, applying in the digital domain modifications to the audio signal based on the one or more other gain components to produce a partly-modified audio signal, and applying in the analog domain modifications to the partly-modified audio signal based on the wideband gain component. Additional loudness modifications other than volume control loudness modifications on the audio signal may also be imposed.

Abstract: An audio crossover system and method is disclosed. An audio system includes two driver circuits, one for each of two audio frequency ranges, e.g., high and low frequency ranges. The driver circuits are designed to provide a combined frequency response curve that has a pronounced midrange attenuation dip, in contrast to prior art designs that attempt to provide a flat response over all frequency ranges.

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

Abstract: A system and method for dynamic bass frequency control, the system comprising: a bandwidth filter coupled to a plurality of switchable electronic components for filtering a frequency range of an input signal; an amplifier coupled to the bandwidth filter and the plurality of switchable electronic components, the amplifier being configured for adjusting amplitude of the filtered frequency range of the input signal based on a configuration of the plurality of switchable electronic components; and a signal level detector coupled to the switchable electronic components for determining signal strength of the input signal, the signal level detector being configured to switch the plurality of switchable electronic components between more than one configuration based on the signal strength of the input signal, each configuration determining the frequency range of the input signal to be filtered by the bandwidth filter and the amplitude adjustment to be made to the filtered frequency range of the input signal by the am

Abstract: A discrete-time audio precompensation filter is designed based on a linear model that describes the dynamic response of a sound generating system at p>1 listening positions. The filter construction is based on providing information representative of n non-minimum phase zeros {Zi} that are outside of the stability region |z|=1 in the complex frequency domain. A causal Finite Impulse Response (FIR) filter, of user-specified degree d, having coefficients corresponding to a causal part of a delayed non-causal impulse response is determined based on the information representative of n non minimum phase zeros. The resulting precompensation filter is determined as the product of at least two scalar dynamic systems, represented by an inverse of a characteristic scalar magnitude response in the frequency domain representing the power gains at the listening positions, and the causal FIR filter designed to approximately invert only non-minimum phase zeros that are safely inverted.

Abstract: An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and/or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

Abstract: A system and method for selectively enhancing an audio signal to make sounds, particularly speech sounds, more distinguishable. The system and method are designed to divide an input auditory signal into a plurality of spectral channels having associated unenhanced signals and perform enhancement processing on a first subset of the spectral channels and not perform enhancement processing on a second subset of the spectral channels. The enhancement processing is performed by determining an output gain for at least the first subset of spectral channels based on a time-varying history of energy of the unenhanced signals associated with each channel in the first subset of the spectral channels and applying the output gain for each of the first subset of the spectral channels to the unenhanced signals to form enhanced signals associated with each of the first subset of the spectral channels.

Abstract: An improved power amplifier system is provided. The power amplifier system includes a programmable digital filter and a power amplifier, each responsive to a plurality of frequency response settings and switching frequency settings, respectively. Each frequency response setting and switching frequency setting is adaptively selected by a processor device to match a bandwidth of an incoming audio signal. The processor device identifies the current bandwidth of an incoming audio signal and adaptively selects a switching rate setting and a frequency response setting based on the current bandwidth. The frequency response setting is selected so as to reduce noise fold over in the power amplifier for a corresponding bandwidth, sampling rate setting, and switching frequency setting.

Abstract: Allpass arrays of arbitrary order are presented. The transducers in the arrays are configured with weights corresponding to the FIR approximation of an allpass filter such that a nearly uniform array response is provided.

Abstract: Systems, methods, and computer program products for displaying audio data are provided. In some implementations, a computer-implemented method is provided. The method includes receiving audio data and displaying a composite image representing the audio data. The composite image combines a first representation and a distinct second representation of the audio data. Each representation is visible in the composite image and the representations appear in the composite image as if laid one over the other.

Abstract: Bass frequencies of audio can be boosted using various techniques and tools. The described techniques and tools can be applied separately or in combination. For example, bass frequencies of audio can be boosted using an integer bass boost filter by receiving user-settable parameters, such as “c” and “s” coefficients, and implementing the integer bass boost filter using a coupled form structure implementation and the user-settable parameters. Bass frequencies of audio can also be boosted using an integer bass boost filter that is configured to use any of plural coupled form structure implementations. Bass frequencies of audio can be also be boosted using a linear combination of an input audio signal and output of a high-pass filter.

Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

Abstract: Sound reproduction is controlled so as to be heard in the optimal state for the hearing function specific to elderly people. A frequency characteristic setting portion for setting the frequency characteristics of an inputted sound signal, and a sound volume setting portion for variable controlling the volume is disclosed. The frequency characteristic setting portion changes a frequency characteristic in which a sound band including a human voice band is emphasized to a frequency characteristic in which the characteristics of a gain in accordance with a frequency gradually becomes flat with an increase in the volume set by the sound volume setting portion.

Abstract: A sound processing apparatus includes a sound input unit configured to input a sound signal, a signal processor having a band-limiting filter for limiting a passing band of the sound signal, and being configured to apply a predetermined signal processing on a sound signal output from the band-limiting filter, and a controller configured to control a band width of the passing band of the band-limiting filter. The controller changes the band width of the passing band according to a level of the sound signal, and makes a speed at the time of expanding the band width faster than a speed at the time of narrowing the band width.

Abstract: A method includes, for each of a number of subbands of a frequency range and for at least first and second frequency-domain signals that are frequency-domain representations of corresponding first and second audio signals: determining a time delay of the first frequency-domain signal that removes a time difference between the first and second frequency-domain signals in the subband. The method includes forming a first resultant signal including, for each of the number of subbands, a sum of one of the first or second frequency-domain signals shifted by the time delay and of the other of the first or second frequency-domain signals; and forming a second resultant signal including, for each of the number of subbands, a difference between the shifted one of the first or second frequency-domain signals and the other of the first or second frequency-domain signals. Apparatus and program products are also disclosed.

Abstract: The invention relates to a method for signal processing of solid-borne sound signals, in particular in motor vehicles, and a corresponding occupant protection system. A first-order high-pass filter is provided as a filter, wherein the ?3 dB cut-off frequency thereof lies between the upper and lower operating frequency. Thus, not only does the signal delay in the filter become significantly lower, but also the frequencies in the operating range become not equal, but rather components below the cut-off frequency are accordingly damped more heavily, whereby triggering cases not only can be detected earlier, but also non-triggering cases can be reliably detected.

Abstract: A multiplier 3 multiplies a square wave signal A(?,t) by a low frequency component signal L(?,t) to generate an even-order harmonic component signal B(?,t). A multiplier 4 multiplies the even-order harmonic component signal B(?,t) by the low frequency component signal L(?,t) to generate an odd-order harmonic component signal C(?,t). An adder 5 adds the even-order harmonic component signal B(?,t) and the odd-order harmonic component signal C(?,t) to generate the harmonic component signal D(?,t).

Abstract: A sound output apparatus includes a mixing unit configured to mix an L (left) channel voice signal and an R (right) channel voice signal to generate an L+WF signal and an R?WF signal containing a mix of a WF(woofer) channel voice signal, and power amplifiers which amplify the L+WF signal and R?WF signal. The outputs of the power amplifiers are BTL-connected to a WF channel speaker. Thus, the mixing unit provided in a previous stage of the power amplifiers generates the WF channel voice signal, which eliminates the necessity for a passive low pass filter in a subsequent power amplifier stage.

Abstract: The present invention relates to an acoustic signal processing apparatus that suppresses auditory noise caused in a difference signal generated by acoustic signals of a plurality of channels, a processing method therefor, and a program. A difference spectrum calculation unit 320 calculates, as a difference spectrum, the absolute difference value between frequency spectra of the left and right channels sent from frequency spectrum generation units 311 and 312. A low-level band determination unit 330 determines a difference spectrum corresponding to a low-level band among difference spectra of all the frequency bands. A substitution spectrum generation unit 350 generates a substitution spectrum for which a difference spectrum is substituted, on the basis of the frequency spectrum of the left channel. A spectrum substitution unit 360 substitutes the difference spectrum corresponding to the low-level band for a substitution spectrum corresponding to the difference spectrum.

Abstract: A method for processing and transducing audio signals. An audio system has a first audio signal and a second audio signal that have amplitudes. A method for processing the audio signals includes dividing the first audio signal into a first spectral band signal and a second spectral band signal; scaling the first spectral band signal by a first scaling factor proportional to the amplitude of the second audio signal; and scaling the first spectral band signal by a second scaling factor to create a second signal portion. Other portions of the disclosure include application of the signal processing method to multichannel audio systems, and to audio systems having different combinations of directional loudspeakers, full range loudspeakers, and limited range loudspeakers.

Abstract: Methods, devices and computer program products facilitate the generation of high quality audio content. The audio content includes upper harmonics that are associated with a bass band of frequencies in order to enhance the perception of bass audio components that cannot be produced by the audio speakers. The upper harmonics are generated and processed in such a way that reduces the computational and memory requirements of the audio processing operations. The processed upper harmonics are combined with the original audio that are properly delayed to enhance the quality of the audio content. This Abstract is provided for the sole purpose of complying with the Abstract requirement rules that allow a reader to quickly ascertain the disclosed subject matter. Therefore, it is to be understood that it should not be used to interpret or limit the scope or the meaning of the claims.

Abstract: Band-pass filter suppresses frequency components of a sound signal that are lower than a low-side cutoff frequency and that are higher than a high-side cutoff frequency. Pitch detection section detects a pitch of the sound signal having been processed by the band-pass filter. Target setting section variably sets a low-side target value lower than the pitch detected by the pitch detection section and a high-side target value higher than the detected pitch. Filter control section causes the low-side cutoff frequency to approach the low-side target value over time and causes the high-side cutoff frequency to approach the high-side target value over time. In this way, a pass band of the band-pass filter can be smoothly variably controlled in accordance with pitch change of the sound signal that is an object of pitch detection.

Abstract: For hearing apparatus and especially for hearing devices a tone balance unit is to be able to be used even under difficult general conditions. Therefore a hearing apparatus with a signal processing device, through which signals are able to be processed in a number of frequency channels, and a control device for controlling the level of the individual frequency channels are provided. The setting device features an unsymmetrical tone balance unit related to a predetermined frequency with which at least one lower and at least one upper channel of the number of frequency channels related to a predetermined frequency is able to be simultaneously unsymmetrically adjusted relative to the predetermined frequency. Thus for example a tone balance unit can also be used with a low feedback limit in the upper frequency range.

Abstract: A circuit for processing sound signals includes an input amplifier for amplifying an input signal, wherein the amplified input signal is supplied to an output amplifier unchanged via a primary path configured as a bypass line and processed via one or several secondary paths configured as a circuit for processing the amplified input signal for amplifying and outputting the processed signal, wherein respective signal portions of the amplified input signal, which are supplied to the output amplifier via the primary path and the secondary path(s), can each be adjusted with an adjusting device, and wherein the secondary path includes a corresponding equalizing filter for filtering a corresponding frequency range with a dynamic circuit for optionally dynamically changing frequency, amplitude and phase depending on the signal and concurrent with a static control.

Abstract: The invention describes an apparatus for software or hardware emulation of electronic audio equipment, which characterizes a non-linear behavior. The invention comprises an analog to digital interface (504) for the input audio signal (502), whose output (506) is communicatively coupled to a dynamic non-linearity (508). The output (514) of this dynamic non-linearity is finally communicatively coupled to an interface (516) producing the output audio signal (518). The dynamic non-linearity consists of a mode switching static non-linear function, where the mode parameter (512) is estimated in a function (510) based on the previous values on the input (506) and output (514) of the dynamic non-linearity.

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

Abstract: A circuit for processing multichannel audio signals, comprises a frequency characteristics correction device and an output device. The frequency characteristics correction device corrects frequency characteristics of an audio signal of a channel including an audio signal component having a predetermined frequency band, of audio signals of a multichannel comprising at least a right channel and a left channel, in accordance with correction characteristics determined based on a head related transfer function. The output device mixes the audio signal component having the frequency characteristics corrected with an audio signal of the right channel and an audio signal of the left channel to generate mixed output audio signals, and outputting the mixed output audio signals as a right channel output audio signal and a left channel output audio signal.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a headliner speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Abstract: An earphone monitoring system is disclosed that provides for user information and control regarding ambient and monitored sound levels.

Abstract: A method is provided for allocating a frequency for wireless audio communications. A request is received for a frequency for wireless audio communications. The request contains the location, the time period, the quantity of required wireless communications paths and/or the available equipment for the wireless audio communications. This request is compared with information about frequency use which is stored in databases. Possible frequency ranges are outputted, and the possible frequency ranges are allocated to transmitting and/or receiving units of the requested audio communications.

Abstract: Systems and methods for audio signal processing are provided. In exemplary embodiments, a filter cascade of complex-valued filters are used to decompose an input audio signal into a plurality of frequency components or sub-band signals. These sub-band signals may be processed for phase alignment, amplitude compensation, and time delay prior to summation of real portions of the sub-band signals to generate a reconstructed audio signal.

Abstract: If the absolute value of the current sample is greater than or equal to an envelope value at the immediately preceding sample, an envelope value at the current sample is made greater than the envelope value at the immediately preceding sample. If the absolute value of the current sample is smaller than the envelope value at the immediately preceding sample and a count value C does not reach a predetermined number N, the count value C is incremented by one and the envelope value at the current sample is held at the envelope value at the immediately preceding sample. If the absolute value of the current sample is smaller than the envelope value at the immediately preceding sample and the count value C reaches the predetermined number N, the envelope value at the current sample is made smaller than the envelope value at the immediately preceding sample.

Abstract: A bass enhancing apparatus and method for enhancing bass include generating harmonics of the bass when an input signal is reproduced using a miniaturized speaker. The bass enhancing method includes extracting a bass component of an input signal, generating harmonics of the extracted bass component, synthesizing the generated harmonic signals and the input signal, and outputting the synthesizing result to an output terminal. The generating of the harmonics includes compressing a dynamic range of an amplitude level of each harmonic component at a predetermined distribution ratio.

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: There is provided an audio signal transmitting apparatus which can communicate with a plurality of external connection apparatuses and transmits audio signals to the external connection apparatuses. The audio signal transmitting apparatus includes an external output unit which outputs the audio signal to the outside, an audio signal transmitting unit which transmits the audio signal to the external connection apparatus being in a communicable state, and an audio output control unit which, when headphone information representing that a headphone function that outputs the audio signal through headphones is valid is received from at least one of the external connection apparatuses which receive the audio signals, performs control to stop an output by the external output unit.

Abstract: A bass and treble audio tone control circuit configured as an integrated circuit, wherein a capacitor for setting a frequency band can be accommodated in an integrated circuit. A LPF extracts a low sound region component SLO from an initial sound signal SIN. A low sound region adjustment circuit adjusts the gain of SLO and generates signal SLT. An inverting circuit inverts SLO and an adding circuit extracts a high sound region component SHO by adding the inverted SLO and SIN. A high sound region adjustment circuit adjusts the gain of SHO and generates a signal SHT. A synthesizing circuit synthesizes SIN with SHT and SLT, and generates an output sound signal SOUT. The LPF is composed of an RC active filter, and the resistance that establishes the cutoff frequency is composed of an equivalent resistance using a switched capacitor circuit.

Abstract: A method and device for frequency compression of audio signals to reduce the occurrence of artifacts. A component of the audio signal having a plurality of frequency channels is shifted from a first frequency channel into a second frequency channel. A dominant instantaneous frequency is determined in the first frequency channel. During shifting or mapping, first the entire first frequency channel, including the dominant instantaneous frequency, is shifted or mapped into the second frequency channel, wherein the dominant instantaneous frequency obtains an intermediate frequency position. A final frequency position for the dominant instantaneous frequency is determined using a predefined compression characteristic in the second frequency channel, starting from the frequency position of the dominant instantaneous frequency in the first frequency channel. Finally, the dominant instantaneous frequency is shifted or mapped from the intermediate frequency position to the final frequency position.

Abstract: Artifacts occurring during frequency compression, in particular in the case of hearing aids, are avoided or reduced. The method compresses the frequency of an audio signal having a fundamental frequency and at least one harmonic. The audio signal is provided in a plurality of frequency channels. The harmonic of the audio signal is shifted or mapped from a first frequency channel of the plurality of frequency channels into a second frequency channel. In addition a frequency which is likewise harmonic with respect to the fundamental frequency is estimated in the second frequency channel, the harmonic being shifted or mapped onto the estimated frequency. As a result the harmonic pattern is preserved in the compressed signal and the artifacts are reduced.

Abstract: Systems and methods for providing bass compensation to correct for uneven bass response are disclosed. An example bass compensation system includes a low pass filter configured to receive an audio signal from an audio source and provide a filtered audio signal, the low pass filter having a roll off of at least 18 dB per octave. The bass compensation system further includes a summing amplifier coupled to the low pass filter and configured to sum the audio signal from said audio source and the filtered audio signal to provide a summed audio signal, wherein the summed audio signal provided by the summing amplifier provides a bass boost at a first frequency and mid bass cut at a second frequency greater than the first frequency.

Abstract: A method of signal communication includes receiving a selection signal, and selecting a selected frequency range used for a multi-carrier signal communication from a set of predetermined frequency ranges for signal communication depending on the selection signal. The set of predetermined frequency ranges includes a first frequency range and a second frequency range including the first frequency range.

Abstract: In one disclosed aspect, dynamic gain modifications are applied to an audio signal at least partly in response to auditory events and/or the degree of change in signal characteristics associated with said auditory event boundaries. In another aspect, an audio signal is divided into auditory events by comparing the difference in specific loudness between successive time blocks of the audio signal.

Abstract: Apparatuses for and methods of carrying out dynamic equalization processing of an audio signal, and apparatuses for and methods of controlling such equalization processing of the audio signal to dynamically adjust the time-varying spectrum of an audio signal to more closely match a user specified target time-invariant perceived audio signal spectrum while preserving the original dynamic range of the audio signal. The dynamic equalization is carried out according to a user-defined spectral profile specified by a control interface that allows a user to define, create, modify and/or apply the user-defined spectral profile.

Abstract: Apparatuses for and methods of carrying out dynamic equalization processing of an audio signal, and apparatuses for and methods of controlling such equalization processing of the audio signal to dynamically adjust the time-varying spectrum of an audio signal to more closely match a user specified target time-invariant perceived audio signal spectrum while preserving the original dynamic range of the audio signal. The dynamic equalization is according to a user-defined spectral profile specified by a control interface that allows a user to define, create, modify and/or apply the user-defined spectral profile.

Abstract: An apparatus is disclosed comprising a transducer, which transducer comprises at least one electrical connection point configured to enable an external electrical connection of the transducer. The transducer further comprises at least one decoupling coil configured to electrically decouple the transducer from the at least one electrical connection point at a frequency above an operating frequency of the transducer. Moreover, a method for assembling the described apparatus is disclosed.

Abstract: There is provided an automatic gain control circuit for volume control receiving at least one audio signal input via either a wired connection or a wireless connection. The circuit includes an output signal regulator receiving a first portion of the at least one audio signal input, with the regulator being able to vary an amplitude of the first portion of the at least one audio signal input when a control signal from a circuit controller is received. The circuit controller may be powered by a DC generator; with the DC generator receiving a second portion of the at least one audio signal input. A signal from the output signal regulator may be passed through an acoustic generator. It is preferable that the first portion of the at least one audio signal input is substantially more than the second portion. It is also preferable that the output signal regulator varies the amplitude of the at least one audio signal input by reducing the amplitude. A corresponding method of volume control is also disclosed.

Abstract: A method and an apparatus for reproducing a sound signal are provided. The method includes generating an output sound signal to be transmitted to speakers by transmitting a first input sound signal through a filter; acquiring magnitude information of the output sound signal; determining frequency response parameters related to frequency responses of the filter based on the magnitude information; and adaptively adjusting coefficients of the filter based on the determined frequency response parameters.

Abstract: A system that includes a direct digital speaker volume control device configured to be coupled to a direct digital speaker. The direct digital speaker includes many pressure producing elements being adapted to generate a sound at a sound pressure level (SPL) and at a given frequency in response to an input signal, without using digital to analog converter. The direct digital speaker inherently exhibits a frequency response throughout its entire frequency range. The direct digital speaker volume control device includes a module for providing few filters each having a distinct cutoff frequency such that each filter exhibits no attenuation below its cutoff frequency and an attenuation response above the filter's cutoff frequency. And a selector for selecting one of the filters according to a selection criterion that depends on a desired volume and frequency of generated sound, and applying the filter to the input signal for generating a filtered signal that fed to the speaker.

Abstract: An energy saving PWM sound device forms a frequency-dependent impedance device implemented on the PWM output of the voice-ICs in series to lower the power consumption and to enhance sound quality of the voice-IC products. The device includes a hollow core body and an elongated conductive element. The core body has a top rim, a bottom rim, and two side rims to define a through channel therewithin. The conductive element winds around the top rim of the core body, wherein two ends of the conductive element are adapted for connecting with the PWM output, such that when the conductive element is electrically conducted, the core body acts as an inductor at lower frequency and a frequency-dependent resistor in series with the inductor at higher frequency for blocking most of PWM carrier frequency so as to reduce the power consumption of the voice-ICs product.

Abstract: A signal processing device includes a filter that is set to frequency characteristics in which a dip existing in an intermediate and high frequency range is smoothed in the frequency characteristics of a virtual characteristic applying filter for applying transfer characteristics of a space transfer path to a sound signal, the space transfer path extending from a virtually set orientation of a sound image to an ear of a listener, an equalizer that forms the dip by cutting a part of the intermediate and high frequency range, and an adjusting unit that adjusts at least a central frequency of the dip. An input signal is passed through the filter and the equalizer.