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: 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: Various embodiments of the present invention are directed to adaptive methods for reducing acoustic echoes in multichannel audio communication systems. Acoustic echo cancellation methods determine approximate impulse responses characterizing each echo path between loudspeakers and microphones within a room and improve performance based on previously determined impulse responses. In particular, the methods adapt to changes in the room by inferring approximate impulse responses that lie within a model of an impulse response space. Over time the method improves performance by evolving the model into a more accurate space from which to select subsequent approximate impulse responses.

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: 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: 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: Described is a unit for amplifying and processing audio signals for driving an electro-acoustic transducer (D), comprising: an input for audio signals; a processor for audio signals (107); an output for a signal for driving said electro-acoustic transducer; and an input for at least one operating quantity of the electro-acoustic transducer. The audio-signal processor is programmed for setting a series of parameters defining a transducer to be emulated, the parameters of which define a model of the transducer to be emulated. The input audio signal is processed on the basis of said at least one operating quantity of the electro-acoustic transducer to obtain a behavior of the electro-acoustic transducer that emulates the transducer defined by said series of parameters set.

Abstract: An audio amplifier is operated in an audio system with optimized source impedance to minimize distortion in a loudspeaker that is paired with the audio amplifier. The audio amplifier provides an amplified audio signal to drive the loudspeaker. A variable output impedance of the audio amplifier is controlled using a feedback control loop to allow negative output impedance at low frequencies that changes to positive output impedance at higher frequencies. The change from negative output impedance to positive output impedance occurs at a determined threshold frequency or determined transitional frequency band.

Abstract: A system includes multiple speakers arranged in a speaker array configuration. The system also includes one or more filters configured to filter audio signals and generate filtered audio signals. The one or more filters are configured to operate using filter coefficients associated with a desired beam pattern to be produced by the multiple speakers. The system further includes at least one amplifier configured to amplify the filtered audio signals and provide the amplified filtered audio signals to the speakers. The one or more filters reside within or are coupled to the at least one amplifier. The system may further include a controller configured to modify at least one of the filter coefficients based on a change in the speaker configuration. The filters may operate independently of a centralized processor, and a centralized processor may not even be required to provide electronic beam forming.

Abstract: A method and apparatus for detecting a malfunctioning speaker are provided. The apparatus detects a speaker abnormality by identifying the state of the speaker when turning the power on or off, switching from standby mode to play mode, or performing testing. The apparatus generates an audio signal having a frequency higher than a particular frequency, outputs the audio signal to the speaker, and determines that the speaker is malfunctioning if the level of a fed-back frequency signal is lower than a reference level. If the speaker is determined to be malfunctioning, the power supply is cut and the danger of a fire starting is accordingly reduced.

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: Various embodiments of the present invention are directed to methods and systems that reduce acoustic echoes in audio signals in accordance with changing conditions at first and second locations that are linked together in a communication system. In particular, one embodiment of the present invention relates to a method for determining an approximate impulse-response vector for canceling an acoustic echo resulting from an audio signal transmitted from the first location to the second location. This method includes forming a trust region within a search space based on computing a recursive specification vector defining the trust region. The method also includes computing a recursive shadow-impulse-response vector that lies substantially within the trust region, and computing the approximate impulse-response vector based on the recursive shadow-impulse-response vector and the recursive specification vector.

Abstract: A dual-sided radio (100) for enhancing a user's experience is provided. The radio includes a primary transducer (110) on an audio-side of the radio that projects a primary sound, a secondary transducer (120) on a data-side of the radio that projects a mid-high frequency sound, a processor (160) that equalizes (200) audio to the primary transducer and the secondary transducer, and a communication module (130) that receives and transmits communication signals containing the audio. The secondary transducer supplements the primary sound with a mid-high frequency sound (404) to compensate for mid-high frequency loss of the primary sound due to diffraction (300).

Abstract: A system for adjusting a response of a loudspeaker system. The system includes a woofer installed in a housing. The woofer includes a woofer cone. A transmitter for transmitting an ultrasonic audio signal towards the woofer cone. A receiver for receiving the ultrasonic audio signal. A detection unit for determining the position of the woofer cone on the basis of a difference between the phase of the reflected ultrasonic audio signal with respect to the ultrasonic audio signal transmitted from the transmitter. A control unit for adjusting a spring constant of the membrane of the loudspeaker system on the basis of the determined position. A method for adjusting a response of a loudspeaker system, to a module to be used in the system. A computer program product including computer code for adjusting a response of the loudspeaker system.

Abstract: A low-cost, real-time solution is presented for compensating memoryless non-linear distortion in an audio transducer. The playback audio system estimates signal amplitude and velocity, looks up a scale factor from a look-up table (LUT) for the defined pair (amplitude, velocity) (or computes the scale factor for a polynomial approximation to the LUT), and applies the scale factor to the signal amplitude. The scale factor is an estimate of the transducer's memoryless nonlinear distortion at a point in its phase plane given by (amplitude, velocity), which is found by applying a test signal having a known signal amplitude and velocity to the transducer, measuring a recorded signal amplitude and setting the scale factor equal to the ratio of the test signal amplitude to the recorded signal amplitude. Scaling can be used to either pre- or post-compensate the audio signal depending on the audio transducer.

Abstract: The invention relates to a hearing instrument for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of operating a hearing instrument and to use of a hearing instrument. The object of the present invention is to provide an alternative scheme for handling acoustic feedback in a hearing instrument.

Abstract: The present invention relates to a method of performing measurements by means of an audio system comprising passive loudspeakers, whereby said measurements loudspeakers for producing sound and at least one of said loudspeakers for measuring said sound. The present invention further relates to an audio system comprising N passive loudspeakers, wherein said audio system further comprises an output stage where each output acts as a combined output channel and a measurement input.

Abstract: A signal processing device includes: a combining section combining a digital feedback signal associated with a movement of a diaphragm of a speaker unit and a digital audio signal; and a control section controlling the level of the digital feedback signal according to a difference between the level of the digital feedback signal and the level of the digital audio signal.

Abstract: A signal processing device includes: a detection unit generating a digital detection signal corresponding to motion of a diaphragm of a speaker to output the digital detection signal; a gain adjustment unit generating a digital feedback signal by multiplying the outputted digital detection signal by again coefficient to output the generated digital feedback signal; a combining unit combining the outputted digital feedback signal with a digital audio signal; a storage unit storing plural gain coefficients; and a control unit performing control so that a given gain coefficient is selected from the plural gain coefficients and that the selected gain coefficient is used for the multiplication.

Abstract: A system and method for Acoustic Echo Cancellation. The system and method include a subband affine projection filter and a variable step size controller configured to cancel an estimated echo from a near-end signal. The system and method also include a divergence detector adapted to reset the subband affine projection filter in response to determining a divergence is occurring. Additionally, the system and method include a double talk detector adapted to transmit a signal to mask an output signal when double talk is detected.

Abstract: In a voice intelligibility enhancement system that controls a gain of a voice signal based on noise power and voice power of the voice signal generated by a voice signal generation unit, it is detected whether the voice power is equal to or greater than a predetermined level, noise power output when the voice power is less than the predetermined level is measured and stored, noise power to be output when the voice power exceeds the predetermined level is estimated to be the stored noise power, and gain of a voice signal is controlled on the basis of the voice power and the estimated noise power.

Abstract: The present invention relates to a method and corresponding system for predicting the perceived spatial quality of sound processing and reproducing equipment. According to the invention a device to be tested, a so-called device under test (DUT), is subjected to one or more test signals and the response of the device under test is provided to one or more means for deriving metrics, i.e. a higher-level representation of the raw data obtained from the device under test. The derived one or more metrics is/are provided to suitable predictor means that “translates” the objective measure provided by the one or more metrics to a predicted perceived spatial quality. To this end said predictor means is calibrated using listening tests carried out on real listeners. By means of the invention there is thus provided an “instrument” that can replace expensive and time consuming listening tests for instance during development of various audio processing or reproduction systems or methods.

Abstract: An amplification circuit for driving an audio signal diffuser that includes a generation circuit of a first pre-charging signal, the generation circuit including an amplifier provided with an input terminal for receiving the first pre-charging signal and provided with an output terminal for providing a second pre-charging signal as a function of the first pre-charging signal, and a decoupling capacitor of the amplifier from the diffuser, the capacitor connected to the output terminal for charging by the second pre-charging signal.

Abstract: An audio system including an audio power amplifier, a transducer electrically connected to the audio power amplifier, an enclosure coupled to the transducer, and a secondary resonant element coupled to the enclosure. An electrical feedback signal representative of the transducer current is negatively fed back to the audio power amplifier to synthesize a positive output impedance.

Abstract: A method and system for determining and compensating for a non-linearity in a hands-free acoustic telecommunication device is disclosed. The method determines a back electromotive force signal induced in a loudspeaker from at least one of a coil voltage, a current signal and estimates of coil resistance and inductance, estimating at least one of a cone position and a cone velocity from the BEMF integrated with respect to time and determining an estimate of an echo value from a series connection of an estimated inverse of a force factor function primitive and the estimated acoustic impulse response; and outputting the estimated echo value.

Abstract: Protection method and system for use with speaker systems. The method and system being suitable for use in any number of environments to protect speakers from overload and other undesirable operating conditions. The method and system optionally being suitable for use in maintaining system sound integrity while protecting the speakers from overload and/or other undesirable operating conditions.

Abstract: Various embodiments of the present invention are directed to adaptive real-time, acoustic echo cancellation methods and systems. One method embodiment of the present invention is directed to reducing acoustic echoes in microphone-digital signals transmitted from a first location to a second location. The first location includes a plurality of loudspeakers and microphones, each microphone produces one of the microphone-digital signals including sounds produced at the first location and acoustic echoes produced by the loudspeakers. The method includes determining approximate impulse responses, each of which corresponds to an echo path between the microphones and the loudspeakers. The method includes determining a plurality of approximate acoustic echoes, each approximate acoustic echo corresponds to convolving a digital signal played by one of the loudspeakers with a number of the approximate impulse responses.

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: An audio signal is processed by passing it through a speaker, the cone of which is mechanically connected to a moving sleeve. A coil is wrapped around the sleeve. There is a metal shaft displaced inside of the sleeve. The shaft is connected to a magnet. Thus, movement of the speaker cone causes the coil to move through a magnetic field which creates a voltage across the coil which is subtracted from the original audio signal to produce a processed output.

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: A feedback reduction system adaptively processes a microphone signal. An adaptive feedback suppression filter circuit processes the microphone signal, and an adaptive feedback compensation filter circuit further processes the microphone signal. The adaptive feedback suppression filter circuit processes the signal based upon output from the adaptive filter compensation filter circuit.

Abstract: A position detection method, system, and computer readable article of manufacture tangibly embodying computer readable instructions for executing the method for detecting the position of a sound source using at least two microphones. The method includes the steps of: emitting a reproduced sound from the sound source; observing the reproduced sound and an observed sound at the microphones; converting the reproduced sound and the observed sound into electrical signals; transforming the signals of the reproduced sound and of the observed sound into frequency spectra by a frequency spectrum transformer apparatus; calculating Crosspower Spectrum Phase (CSP) coefficients of the frequency spectra of the signals by a CSP coefficient calculator apparatus; and calculating distances between the position of the sound source and the positions of the microphones based on the calculated CSP coefficients by a distance calculating apparatus, thereby detecting the position of the sound source.

Abstract: Distortion caused in the voltage of an audio signal due to a back electromotive force generated due to loads in a speaker is detected, the detected distorted voltage is eliminated from the voltage of the audio signal, and the audio signal without the distorted voltage is amplified and given to the speaker.

Abstract: The present invention is embodied in a computer-readable program in a computer-readable medium for upgrading a video conference system, the computer-readable program comprising acoustic echo canceling control software having an application programming interface. The acoustic echo canceling control software is implemented on a computer system that operates the video conference system and macros are configured to couple the acoustic echo canceling control software to hardware components of the video conference system and to interface with the application programming interface. The macros are user configurable for providing real time adjustments of echo canceling runtime parameters of the hardware components during a video conference session.

Abstract: The invention relates to a method and a system for adapting a loudspeaker to a specific listening position relative to the loudspeaker according to which method and system the acoustic power radiated by the loudspeaker is corrected by means of a correction filter inserted in the signal path through the loudspeaker, the response of said correction filter being determined by comparison between the a quantity characterising the radiated acoustic power measured at an actual listening position and a similar quantity measured at a reference listening position. According to a specific embodiment of the invention said characterising quantities are the radiation resistances measured at the actual listening position and the reference listening position respectively.

Abstract: A compound loudspeaker comprises an acoustically radiating first diaphragm and an acoustically radiating second diaphragm. The first and second diaphragms are substantially coaxial and at least part of the second diaphragm is situated radially outwards of the first diaphragm. There is a gap situated between the first and second diaphragms, and a seal is provided in the gap, thereby preventing or hindering the passage of air through the gap. By providing the seal, the invention solves the problem of audible turbulent airflow through the gap.

Abstract: A speaker assembly arrangement for a vehicle is provided that includes a speaker unit including a speaker cabinet defining a first volume and a body part of a vehicle including a hollow member defining a second volume. The speaker cabinet is acoustically coupled to the hollow member by a press-fit coupling having a press-fit direction and mounted on the body part by at least one fastener having a fastening direction substantially perpendicular to the press-fit direction. The body part comprises a first guide arranged and configured to slidably engage with a second guide arranged on the speaker cabinet so that force on the speaker cabinet in the fastening direction is at least in part redirected to the press-fit direction by the slidable engagement of the first guide and the second guide.

Abstract: A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

Abstract: A reproducing apparatus includes: a connecting part which is connected to an audio signal output apparatus through the three pole terminal; a storage part which stores content data including first and second channel audio data; a reproducing part which reproduces the content data; a first converting part which converts the first channel audio data included in the content data reproduced by the reproducing part into the first channel audio signal; a second converting part which converts the second channel audio data included in the content data reproduced by the reproducing part into the second channel audio signal; and a measuring part which measures at least any one of an amount of sound leakage in the second output unit caused by outputting the first channel audio signal and an amount of sound leakage in the first output unit caused by outputting the second channel audio signal.

Abstract: A system for reducing an echo and/or a residual echo in a microphone output signal includes an echo compensation filter configured to receive audio input signals and generate an estimated echo signal. A speech activity detector is configured to detect speech of a local speaker, and a combining circuit is configured to generate an echo compensated signal by subtracting the estimated echo signal from the microphone output signal. A residual echo reduction circuit is configured to suppress the residual echo in the echo compensated signal based on the detected speech activity, and output an echo suppressed output signal. The echo reduction system may reduce local background noise and residual echoes in a microphone output signal so that local speech may be transmitted free from undesirable signals.

Abstract: Disclosed is a method for equalizing a first loudspeaker positioned in a room in order to compensate for an influence of the room, the method comprising the steps of 1) measuring a listening position transfer function from electrical input of the first loudspeaker to a sound pressure at a listening position in the room, 2) determining a global transfer function representing a spatial average of sound pressure level in the room generated by the first loudspeaker, 3) determining an upper gain limit as a function of frequency, the upper gain limit being based on an inverse of the global transfer function, 4) determining an equalizing filter based on an inverse of the listening position transfer function, wherein a gain of the equalizing filter is limited to a maximum gain in accordance with the upper gain limit, and 5) equalizing the first loudspeaker according to the equalizing filter.

Abstract: A system reduces echoes in an audio system by de-correlating audio signals. Adaptive filters provide compensation based on the de-correlated signals. A controller controls the de-correlation based upon the adaptation state of the adaptive filters.

Abstract: A system automatically determines an equalizing filter characteristic for a communication system within a vehicle. The communication system includes a loudspeaker and a microphone or microphone array. The system transmits a predetermined test signal through the loudspeaker and receive the test signal through the microphone or microphone array. Based on the predetermined test signal and the received test signal, a transfer function is developed. The equalizing filter characteristic is then developed from the transfer function.

Abstract: A method is provided for creating a series of digital signal processing (DSP) filters to improve the transient response of a loudspeaker, wherein the loudspeaker is formed of multiple components. The method includes generally six steps. The first step involves identifying a substantially linear, time-invariant, and spatially-consistent loudspeaker mechanism causing transient response distortion. The second step involves characterizing the identified mechanism. The third step involves determining the characterized mechanism's two-port response. The fourth step involves establishing a target response for the characterized mechanism. The fifth step involves calculating an ideal filter to achieve the target response. The sixth step involves designing a cost-reduced filter based on the ideal filter.

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: The loudspeaker device according to the present invention comprises a loudspeaker; a feedforward processing section for performing feedforward processing on an electric signal to be inputted to the loudspeaker based on a preset filter coefficient so that non-linear distortion which occurs from the loudspeaker is removed; and a feedback processing section for detecting vibration of the loudspeaker, and performing feedback processing on an electric signal concerning the vibration with respect to the electric signal to be inputted to the loudspeaker. The feedback processing section performs feedback processing on the electric signal concerning the vibration so that the non-linear distortion which occurs from the loudspeaker is removed and so that a frequency characteristic concerning the vibration of the loudspeaker becomes a predetermined frequency characteristic.

Abstract: A capacitive transducer circuit includes a capacitive transducer having first and second electrodes. The first and second electrodes are biased by respective first and second bias voltages. An amplifier is connected to receive a first analog signal on an input terminal, the first analog signal being generated by the capacitive transducer, and to generate a second analog signal on an output terminal. A digital feedback circuit is connected between the output terminal of the amplifier and the input terminal of the amplifier. The digital feedback circuit is configured to provide one of said first or second bias voltages. A switched capacitor filter circuit may be arranged between the voltage source and the transducer and may be arranged to filter the output of the voltage source.

Abstract: The invention concerns a method and device for actively correcting acoustic properties of a listening zone (27) of an acoustic space (1), comprising means for converting a signal to be reproduced (2, 4) in an acoustic space (1), means for attenuating resonance including at least means for measuring the perturbed sound signal (15), means for processing an electric signal (6) and at least one secondary sound reproduction source (8).

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: Various embodiments of the present invention are directed to methods and systems that reduce acoustic echoes in audio signals in accordance with changing conditions at first and second locations that are linked together in a communication system. In one embodiment of the present invention, a first digital signal encoding sounds produced at the first location is output from the first location, and a second digital signal encoding an acoustic echo and sounds produced at the second location is output from the second location. The method computes a control state that depends on the signals transmitted between the first and the second locations and computes an approximate acoustic echo based on the control state.