Abstract: An example method of operation may include receiving audio data at one or more microphones disposed in a corresponding plurality of network devices, identifying amplitude values of the audio data at each of the plurality of network devices, and each of the amplitude values identified at each of the plurality of network devices are different from each other of the amplitude values, determining at each of the plurality of network devices a location of the audio data based on a direction and amplitude of the received audio data, modifying the audio data for output via a plurality of loudspeakers disposed in each of the plurality of network device, and outputting, via the plurality of loudspeakers, the modified audio data, and each loudspeaker outputs different versions of the modified audio data.

Abstract: A device includes one or more processors configured to execute instructions to determine a first phase based on a first audio signal of first audio signals and to determine a second phase based on a second audio signal of second audio signals. The one or more processors are also configured to execute the instructions to apply spatial filtering to selected audio signals of the first audio signals and the second audio signals to generate an enhanced audio signal. The one or more processors are further configured to execute the instructions to generate a first output signal including combining a magnitude of the enhanced audio signal with the first phase and to generate a second output signal including combining the magnitude of the enhanced audio signal with the second phase. The first output signal and the second output signal correspond to an audio zoomed signal.

Abstract: At least one exemplary embodiment is directed to a method of generating a hearing impairment compensation function to process audio reproduced by an earphone device. The compensation function includes processing the audio to compensate for frequency sensitivity to improve speech intelligibility.

Abstract: The present disclosure relates to an integrated circuit connectable to a microelectromechanical system (MEMS) transducer. The MEMS transducer is configured to generate a transducer audio signal in response to sound. The integrated circuit comprises a digital scrambling circuit coupled to a data communication interface. The digital scrambling circuit is configured to convert a digital audio stream, representative of the transducer audio signal, into a corresponding scrambled data stream. The integrated circuit additionally comprises a data bus interface coupled to the digital scrambling circuit and configured to output the scrambled data stream.

Abstract: A microphone assembly includes a transducer element and a processing circuit. The processing circuit includes an analog-to-digital converter (ADC) configured to receive, sample and quantize a microphone signal generated by the transducer element to generate a corresponding digital microphone signal. The processing circuit includes a feedback path including a digital loop filter configured to receive and filter the digital microphone signal to provide a first digital feedback signal and a digital-to-analog converter (DAC) configured to convert the first digital feedback signal into a corresponding analog feedback signal. The processing circuit additionally includes a summing node at the transducer output configured to combine the microphone signal and the analog feedback signal.

Abstract: A MEMS acoustic transducer has: a detection structure, which generates an electrical detection quantity as a function of a detected acoustic signal; and an electronic interface circuit, which is operatively coupled to the detection structure and generates an electrical output quantity as a function of the electrical detection quantity.

Abstract: Sensor data from a sensor system of a mobile device may be used for determining a level of pressure exerted by a user on the mobile device. The sensor system may include one or more types of sensors, such as a microphone and one or more inertial sensors. The inertial sensors may include one or more gyroscopes and/or accelerometers. Based on the inertial sensor data, it may be determined whether and/or how the mobile device is being held. A process for determining a level of pressure exerted by a user on the mobile device may be adapted based, at least in part, on whether and/or how the mobile device is being held. The pressure-determining process may be adapted according to various other criteria, such as a position of a touch target in a display, ambient noise levels, etc.

Abstract: A novel ribbon microphone incorporates rounded-edge magnet motor assembly, a backwave chamber, and a phantom-powered JFET circuit. In one embodiment of the invention, one or more novel rounded-edge magnets may be placed close to a ribbon of the ribbon microphone, wherein the one or more novel rounded-edge magnets reduce or minimize reflected sound wave interferences with the vibration of the ribbon during an operation of the ribbon microphone. Furthermore, in one embodiment of the invention, a novel backwave chamber operatively connected to a backside of the ribbon can minimize acoustic pressure, anomalies in frequency responses, and undesirable phase cancellation and doubling effects. Moreover, in one embodiment of the invention, a novel phantom-powered JFET preamplifier gain circuit can minimize undesirable sound distortions and reduce the cost of producing a conventional preamplifier gain circuit.

Abstract: A hearing protector has two closed muffs (2) which sealingly abut against the wearer's head, so that there is formed a closed space in each muff (2) and about the user's ear. A loudspeaker (4) is disposed interiorly in the muff and has a membrane (5) whose one side is turned to face towards the user's ear, and whose opposing side is turned to face towards a closed volume which is defined by the membrane and an enclosure. In order to improve the base reproduction in listening to music, the interior of the enclosure is, via at least one hole or one duct, in communication with the ambient surroundings outside the closed space of the muff (2). In one preferred embodiment, the enclosure comprises a capsule (7) which is integrated part of the loudspeaker (4).

Abstract: The present disclosure provides an audio quantization coding and decoding device and a method thereof. In the method, before a quantization coding process is performed on a digital signal, the signal is pre-processed, the digital signal is split into multiple frames based on positive and negative half periods of the signal, and all audio data between two adjacent zero-crossing points belongs to the same positive and negative half periods, so as to have the same sign-bit. A pre-processing module groups the numeric data belonging to the same positive and negative half periods into the same frame. When coding, an audio quantization coding module only needs to record a sign-bit of the frame at a head of the frame, so the sign-bit of each batch of voice data in the frame may be omitted to reduce a data amount or improve a resolution of each batch of voice data.

Abstract: An acoustic feedback suppression apparatus includes an amplitude measurement section for measuring the amplitude of a sound signal input from a microphone, an amplitude attenuation section for attenuating the amplitude of a sound signal output to a speaker, and an acoustic feedback determination section for determining whether or not acoustic feedback is occurring based on the sound signal input from the microphone. If it is determined that acoustic feedback is occurring, a sound-deadening control section enables sound-deadening control so that the amplitude of the sound signal output to the speaker is attenuated to a sound-deadening level. A sound-deadening release section releases the sound-deadening control if the amplitude of the sound signal input from the microphone is more than specified threshold amplitude and if it is determined that acoustic feedback is not occurring while the sound-deadening control is conducted.

Abstract: The present application provides a system, method and non-transitory computer readable medium that provides a means of using a vehicle horn as an inexpensive user input interface to serve the function of a simple push button switch for an in-vehicle device or system. The use of the vehicle horn as a user input interface to an in-vehicle device or system is novel. The description of example embodiments illustrates application details that take advantage of the properties of the vehicle horn sound to allow efficient processing that can be implemented on a low cost processor.

Abstract: A method for controlling adaptation of a feedback suppression filter in a hearing aid comprises the step of suspending the adaptation of the feedback suppression filter when the level of a hearing aid reference signal is below a predetermined threshold. The invention further provides a hearing aid with a feedback suppression filter and with means for suspending adaptation of the feedback suppression filter.

Abstract: A processing chip for a digital microphone and related input circuit and a digital microphone are described herein. In one aspect, the input circuit for a processing chip of a digital microphone includes: a PMOS transistor, a resistor, a current source, and a low-pass filter. The described processing chip possesses high anti high-frequency interference capabilities and the described input circuit possesses high high-frequency power supply rejection ratio.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.

Abstract: Near-end equipment for a communication channel with far-end equipment. The near-end equipment includes at least one loudspeaker, at least two microphones, a beamformer, and an echo canceller. The communication channel may be in one of a number of communication states including Near-End Only state, Far-End Only state, and Double-Talk state. In one embodiment, when the echo canceller determines that the communication channel is in either the Far-End Only state or the Double-Talk state, the beamformer is configured to generate a nearfield beampattern signal that directs a null towards a loudspeaker. When the echo canceller detects the Near-End Only state, the beamformer is configured to generate a farfield beampattern signal that optimizes reception of acoustic signals from the near-end audio source. Using different beamformer processing for different communication states allows echo cancellation processing to be more successful at reducing echo in the signal transmitted to the far-end equipment.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: The subject disclosure is directed towards a noise adaptive beamformer that dynamically selects between microphone array channels, based upon noise energy floor levels that are measured when no actual signal (e.g., no speech) is present. When speech (or a similar desired signal) is detected, the beamformer selects which microphone signal to use in signal processing, e.g., corresponding to the lowest noise channel. Multiple channels may be selected, with their signals combined. The beamformer transitions back to the noise measurement phase when the actual signal is no longer detected, so that the beamformer dynamically adapts as noise levels change, including on a per-microphone basis, to account for microphone hardware differences, changing noise sources, and individual microphone deterioration.

Abstract: A pseudo noise superimposing unit superimposes a pseudo noise (M-sequence) to an audio signal picked up by a microphone and outputs the superimposed signal to an amplifying system. An calculating unit calculates a correlation value between the audio signal picked up by the microphone and the pseudo noise. The calculating unit estimates a gain of a closed loop based on the correlation value. A gain control unit suppresses a gain of the audio signal based on the estimated gain of the closed loop.

Abstract: Systems, methods, and other embodiments associated with selective notch filtering for howling suppression are described. According to one embodiment, an apparatus includes a howling detector that detects howling by performing a time domain analysis of speech signals to identify a speech signal that may be exhibiting howling and performing frequency domain analysis to confirm that the speech signal is exhibiting howling. The apparatus also includes a suppression selector configured to select a suppression technique, a signal processor configured to process the speech signal according to a selected suppression technique. The apparatus outputs, without suppression-related processing, speech signals that are not identified based on the time domain analysis.

Abstract: A method and apparatus for receiving and playing a signal in a radio receiver to suppress microphonic feedback are provided by alternately pitch shifting a received audio signal. The pitch of the received audio signal is alternately shifted up and then down, repeatedly over successive intervals of the audio signal, to produce a pitch swing signal which is then played over a speaker. The alternating pitch shifting prevents the buildup of regenerative feedback normally caused by acoustic vibrations coupling into the radio receiver.

Abstract: There is provided an audio amplifier circuit (200), comprising: an input, for receiving an input signal; an amplifier (205) for receiving a modified input signal and outputting an amplified signal; and a feedback loop (245) comprising digital circuitry for feeding back the amplified signal and combining a feedback signal with the input signal to generate the modified input signal. The feedback loop (245) comprises: a low-pass filter (220) for filtering the amplified signal; a comparator (225) for comparing the filtered signal with a threshold value and outputting a comparison signal; and an integrator (230) for integrating the comparison signal and outputting the feedback signal. According to another embodiment, the feedback loop comprises an integrator and a sigma-delta modulator.

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: An autobias vehicular microphone system (300) includes a microphone (301) which uses an amplifier (306) for amplifying an output of the microphone. A first feedback path (308) provides an amplifier output signal to the amplifier input for providing amplifier linearity, and a second feedback path (305) is used for providing bias to a voltage reference (303). The voltage reference (303) operates to provide an autobias to the amplifier (306) based upon amplifier load-ing. By holding the bias point to a constant voltage, a constant clip level can be maintained depending on varying load conditions of electronic devices (307, 309, 311) using the microphone (301). Additionally, one or more switches can be used to vary the bias point which can be interpreted to control functionality of the electronic devices (307, 309, 311).

Abstract: A method of setting a parameter for an audio apparatus including a delay processing unit which applies a delaying process and an amplifying process on an input audio signal in accordance with the set parameter, includes: performing a measuring process including: causing an outputting device to output measurement sound; and measuring a first impulse response at a sound receiving point; performing a specifying process including: analyzing the measured first impulse response; calculating a second impulse response at the sound receiving point when the outputting device outputs sound in case where a parameter is set in the audio apparatus and an audio signal indicative of the measurement sound is input to the audio apparatus; and specifying a parameter in accordance with the calculated second impulse response; and performing a setting process of setting the specified parameter in the delay processing unit.

Abstract: A method for a multi microphone noise reduction in a complex noisy environment is proposed. A left and a right noise power spectral density for a left and a right noise input frame is estimated for computing a diffuse noise gain. A target speech power spectral density is extracted from the noise input frame. A directional noise gain is calculated from the target speech power spectral density and the noise power spectral density. The noisy input frame is filtered by Kalman filtering method. A Kalman based gain is generated from the Kalman filtered noisy frame and the noise power spectral density. A spectral enhancement gain is computed by combining the diffuse noise gain, the directional noise gain, and the Kalman based gain. The method reduces different combinations of diverse background noise and increases speech intelligibility, while guaranteeing to preserve the interaural cues of the target speech and directional background noises.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: A method for reducing howling in a communication system containing collocated mobile devices is presented. In a transmitter, an audio signal is received at a microphone. Acoustic feedback is removed from the audio signal and the resulting signal is encoded and transmitted either using direct or trunked mode operation to a receiver. The encoded signal is decoded at the transmitter, in addition to at the receiver, and fed back to an echo canceller with sufficient delay to account for substantially the entirety of a loop delay from encoding of the audio signal to reception of the acoustic feedback at the microphone to enable removal of the acoustic feedback. An estimate of the acoustic feedback is used to initially remove the acoustic feedback, the error being fed back to the processor to adaptively change the signal being subtracted from the audio signal to better reduce the acoustic feedback.

Abstract: A communication device including multiple microphones is provided. The communication device includes at least two microphones. The communication device further includes a sidetone feedback notifier for producing a notification signal. The sidetone feedback notifier is coupled to the microphones. The notification signal is based on the combination of, a first input audio signal provided for by a first microphone, and a second input audio signal provided for by a second microphone. The sidetone feedback notifier is coupled to a notification device for providing a feedback signal to a user based on the notification signal.

Abstract: In one embodiment, a method comprising receiving at a microphone located at a first location audio received from plural speakers, the audio received at a first amplitude level; and responsive to moving the microphone away from the first location to a second location, causing adjustment of the audio provided by the plural speakers to target the first amplitude level at the microphone.

Abstract: A sound processing device includes: a plurality of sound input units; a detecting unit for detecting a frequency component of each sound input to the plurality of sound signal unit, the each sound arriving from a direction approximately perpendicular to a line determined by arrangement positions of two sound input units among the plurality of sound input units; a correction coefficient unit for obtaining a correction coefficient for correcting a level of at least one of the sound signals generated from the input sounds by the two sound input units so as to match the levels of the sound signals with each other based on the sound of the detected frequency component; a correcting unit for correcting the level of at least one of the sound signals using the obtained correction coefficient; and a processing unit for performing a sound process based on the sound signal with the corrected level.

Abstract: The invention provides an amplifier circuit of a capacitor microphone of which the noise resistance against noise of a supply voltage is enhanced. In an amplifier circuit of a capacitor microphone of the invention, while a noise component of a supply voltage is applied to one inversion input terminal of an operational amplifier of an amplification portion through a parasitic capacitor existing between an external power supply wiring and an external wiring that are adjacent to each other, the problem noise component of the supply voltage is applied to the other non-inversion input terminal by capacitive coupling to an internal power supply wiring. Therefore, the noise component is cancelled at the operational amplifier.

Abstract: An electrical apparatus includes a frame, a speaker connected to the frame, a digital signal processor in communication with the speaker to receive audio data and control data to control the speaker, the digital signal processor connected to the frame, and a lamp base coupler electrically connected to the speaker and receiver, the lamp base coupler detachably connectable to a power source, when the power source is present. A method of steering the diffused sound field includes, broadcasting at least one calibration audio signal through a plurality of speakers (M) in an audio system, receiving the calibration audio signal in a plurality of microphones spaced apart and positioned about at a listening position, and calculating respective relative speaker placement angles relative to the listening position between each of the plurality of speakers in response to receipt of the calibration audio signal in the plurality of microphones.

Abstract: There is disclosed a microphone, a circuit, and a method. A microphone capsule may include an electret microphone and a field effect transistor (FET). A floating DC voltage source may have a first end connected to a drain terminal of the electret microphone capsule and a second end. A load resistor may be connected between the second end of the floating DC voltage source and a source terminal of the electret microphone capsule. A voltage follower may have an output connected to the source terminal of the electret microphone capsule and the first end of the floating DC voltage source. A coupling capacitor may couple an audio signal from the source terminal of the electret microphone capsule to an input of the voltage follower.

Abstract: There is provided an apparatus for enabling karaoke. The apparatus includes a casing for the apparatus; a microphone array incorporated within the casing, the microphone array being concealed from a user by the casing; a controller coupled to the microphone array to at least process audio signals from the user input at the microphone array; a data storage device coupled to the controller, the data storage device being for storing pre-recorded songs used for karaoke, and for storing songs as sung by the user; and an image capturing device coupled to the controller, the image capturing device being for capturing images of the user while the user is singing, with the captured images of the user being stored on the data storage device.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: According to one embodiment, a pickup signal processing apparatus includes microphones, a sound determining unit, a signal level calculating unit, a setting unit, and a calculating unit. The sound determining unit determines whether pickup signals picked up by the microphones are signals from a neighboring sound source or a background noise signal. The signal level calculating unit calculates the signal levels for the microphones. The setting unit sets a gain value of at least one microphone and reduces a difference between the signal levels for the microphones on the basis of the signal levels for the microphones, when determined that the pickup signal is the background noise signal. The calculating unit multiplies the pickup signal of the at least one microphone by the gain value set by the setting unit.

Abstract: An audio system for a vehicle has at least one source of audio signals. A respective directional loudspeaker array is mounted at each seat position and coupled to the at least one source. The at least one source includes a microphone that detects speech from an occupant of the first seat position. Processing circuitry receives signals from the microphone that correspond to the detect speech and drives each second respective loudspeaker array at the other seat positions to radiate acoustic energy corresponding to the detected speech.

Abstract: The invention concerns in a first aspect an audio system comprising a microphone, audio signal processing means, an output transducer and means for detecting a possible feedback tone and the corresponding frequency of the feedback tone in the audio system between the output transducer and the microphone. According to the invention means for counteracting feedback are provided. Further, means are provided for changing the phase of the audio signal at a given frequency. In a further aspect, a binaural hearing aid system comprising first and second hearing instruments according to the first aspect, each hearing instrument comprising transceiver circuitry allowing an exchange of signals between the two hearing instruments, and wherein the binaural hearing aid system is adapted to provide that a phase change introduced in the first audio signal by a controller of the first hearing instrument is or can be introduced in the second audio signal of the second hearing instrument via said transceiver circuitry.

Abstract: A howling suppression device that can reduce quality deterioration of processed sound includes: a delay unit delaying the input signal to output the delayed input signal as a reference signal; a signal separation unit including an adaptive filter extracting a periodic signal component from the reference signal by adaptively updating a filter coefficient; a howling detection unit detecting an occurrence of howling using at least a signal of the periodic signal component output from the adaptive filter; and a howling suppression unit. The howling suppression unit includes: a suppression filter obtaining the updated filter coefficient from the adaptive filter with timing when the howling detection unit detects the occurrence of the howling, to extract the periodic signal component from the reference signal based on the filter coefficient; and a subtractor subtracting the periodic signal component from the input signal so as to output a signal obtained by the subtraction.

Abstract: A method for detecting whistling in an audio system includes determining an average frequency of an input signal of the audio system, sampling the input signal in consecutive blocks of at least one sample, wherein the average frequency is determined blockwise, and determining whether feedback related whistling is present in the input signal of the audio system by evaluating a stability of the average frequency, wherein the evaluation of the stability of the average frequency comprises: determining a difference of two values of the determined average frequency for two blocks, and comparing the determined difference to a first threshold value.

Abstract: A method and apparatus for receiving and playing a signal in a radio receiver to suppress microphonic feedback are provided by alternately pitch shifting a received audio signal. The pitch of the received audio signal is alternately shifted up and then down, repeatedly over successive intervals of the audio signal, to produce a pitch swing signal which is then played over a speaker. The alternating pitch shifting prevents the buildup of regenerative feedback normally caused by acoustic vibrations coupling into the radio receiver.

Abstract: Method and apparatus for signal processing an input signal in a hearing assistance device to avoid entrainment, the hearing assistance device including a receiver and a microphone, the system comprising using a gradient adaptive lattice filter including one or more reflection coefficients to measure an acoustic feedback path from the receiver to the microphone of the hearing assistance device.

Abstract: A circuit includes a microphone circuit, an audio processing module, a digital audio processing module, and an active noise reduction (ANR) circuit. The microphone circuit receives acoustic vibrations and generates an audio signal therefrom. The audio processing module generates a representation of the audio signal. The digital audio processing module compensates the representation of the audio signal based on hearing compensation data to produce a hearing compensated audio signal. The ANR circuit receives the hearing compensated audio signal and an ANR signal. The ANR circuit further functions to adjust the hearing compensated audio signal based on the ANR signal to produce an output audio signal, wherein the ANR signal is generated based on the output audio signal.

Abstract: Implementations related to echo cancellation are depicted and described herein.

Abstract: An audio processing system is provided for controlling the acoustics of a loudspeaker-room system. The loudspeaker-room system having a listening room and loudspeakers located in said listening room, and transfer functions with linear and non-linear components. The audio processing system comprises a compensator with a transfer function for obtaining at least two compensated signals from the input signals. The transfer functions of the compensator may include linear and non-linear components and are inverse to the transfer functions of the loudspeaker-room system to the extent that a desired overall transfer function is established.

Abstract: An audio system for an aircraft passenger cabin and a method for controlling such an audio system for providing passengers with information programs, in which multiple loudspeakers are each disposed at a predefined loudspeaker location in the aircraft cabin, and an audio signal having at least one audio channel is reproduced via the loudspeakers wherein the at least one audio channel is output by a central processing unit to the multiple loudspeakers in the aircraft cabin via a bus system having multiple distribution units and multiple cabin units.