Abstract: A sound reinforcement system which enables handsfree and high-quality sound reinforcement without requiring a person who is speaking to move to a microphone or move a microphone. At least one microphone and a plurality of speakers are arranged in a room. A speaker output adjusting section outputs sound picked up by the microphone to the plurality of speakers at predetermined levels.

Abstract: A speaker system utilizes input from a transducer that receives an acoustical signal produced by a speaker that is separate from the speaker system. The acoustical signal is amplified and drives a speaker of the speaker system. The speaker system is thereby provided with the necessary electrical audio signal without the speaker system being wired into any existing sound system wiring. In automobiles, the transducer is in proximity to a speaker wired into the automobile's audio system, and the transducer obtains the acoustical signal to produce the electrical audio signal used by the speaker of the speaker system so that no access to high or low level electrical audio signals of the audio system of the automobile is necessary. Additionally, the speaker system may employ a power socket plug that is electrically coupled to the power input of the amplifier and that may be plugged into a power socket such as those typical of most vehicles to provide electrical power to the amplifier of the speaker system.

Abstract: An acoustics correcting apparatus includes: a measurement signal supplying section; first and second collecting sections; a first distance calculating section; a second distance calculating section; a position information calculating section; an acoustics measuring section; a virtual sound image coefficient selecting section; a correction characteristic calculating section; a virtual sound image localization processing section; and an acoustics correcting section.

Abstract: A procedure and device for linearizing the characteristic curve of a vibration signal transducer such as a microphone that includes collecting signals, transmitting the signals, extracting information from the signals, dephasing such information by 180 degrees compared to the initial signals and taking the algebraic sum of the initial signals and dephased information.

Abstract: Disclosed is a tuning sound (TS) feedback apparatus. The tuning sound (TS) feedback apparatus includes: a TS pick-up casing which is disposed and fixed inside or near the hearing organ of a sound engineer who manages tuning operations for sound output from a sound output terminal of a sound device having an equalizer; a sound/electric conversion element which is included in the TS pick-up casing, picks up the TS output from the sound output terminal of the sound device, and converts the picked-up TS into an electrical signal; and a TS output module which is included in the TS pick-up casing, converts the TS converted into the electrical signal by the sound/electric conversion element into wired or wireless communication data to be used for external communication, and transmits the data to the sound device.

Abstract: A system for compensating and driving a loudspeaker includes an open loop loudspeaker controller that receives and processes an audio input signal and provides an audio output signal. A dynamic model of the loudspeaker receives the audio output signal, and models the behavior of the loudspeaker and provides predictive loudspeaker behavior data indicative thereof. The open loop loudspeaker controller receives the predictive loudspeaker behavior data and the audio input signal, and provides the audio output signal as a function of the audio input signal and the predictive loudspeaker behavior data.

Abstract: The invention concerns an audio system comprising a microphone, audio signal processing means, an output transducer and mans 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, mans are provided for changing the phase of the audio signal at a given frequency.

Abstract: A device and a method for integrating 3D sound effect processing and active noise control are proposed. A digital signal processor incorporates an artificial reverberator and a 3D spatial audio processor into an audio module. The audio signal is presented via an earphone. Next, a microphone embedded in the vicinity of the loudspeaker inside the headset is used to sense an external noise while playing, and feed it back to an active noise controller, which generates an anti-noise to eliminate the external noise. Therefore, the signal to noise ratio can be increased and the 3D sound field effect can be significantly enhanced. In addition, a head-related transfer function is more efficiently implemented on the basis of an interaural transfer function in the spatial audio processing to reduce the filter order lower and hence the computation loading.

Abstract: The present application relates to a sound reduction device as well as to a corresponding method. The sound reduction device comprises a sound pickup for measuring an occurring error signal of a primary sound wave of the noise source and of a secondary sound wave of a narrow-band electroacoustic transducer. This error signal may be transmitted to a control unit, which receives a reference signal of the noise source and generates a control signal which is adapted to change the mechanical values of the electroacoustic transducer.

Abstract: A robotic surgical system has a robot arm holding an instrument for performing a surgical procedure, and a control system for controlling movement of the arm and its instrument according to user manipulation of a master manipulator. The control system includes a filter in its forward path to attenuate master input commands that may cause instrument tip vibrations, and an inverse filter in a feedback path to the master manipulator configured so as to compensate for delay introduced by the forward path filter. To enhance control, master command and slave joint observers are also inserted in the control system to estimate slave joint position, velocity and acceleration commands using received slave joint position commands and torque feedbacks, and estimate actual slave joint positions, velocities and accelerations using sensed slave joint positions and commanded slave joint motor torques.

Abstract: A user/machine interface comprising a panel having a surface, the panel being capable of supporting bending waves, a touch-sensitive input device associated with the surface, and means including a force transducer for providing force feedback to the input device. The force is in the form of pulses to the panel, the pulses being in the form of a modulated signal shaped as a damped sinusoid whereby a button click sensation is provided to the user's finger tip. The modulated signal may be produced by a narrow-band sine wave having a carrier frequency in the range 150 to 750 Hz and being of a duration of at least 10 ms.

Abstract: Hearing aids with microphone and telephone coil are to be made simpler and more convenient. For this purpose it is provided to use an adaptive filter to compensate acoustic and electromagnetic feedback. In order to allow for the propagation delay differences, a delay element is connected downstream of the telephone coil. The microphone and telephone coil signals can be individually weighted with the factors a and b so that mixed mode is also possible.

Abstract: A method and apparatus for loudspeaker equalization are described. In one embodiment, the method comprising generating a set of parameters using an invertible, non-linear system based on input audio data and output data corresponding to a prediction of an output of a loudspeaker in response to the input data, and controlling an exact non-linear inverse of the non-linear system using the set of parameters to output a predistorted version of the input data.

Abstract: A method of comparison between pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of (n) microphones mi and (p) speakers hpj for the control of said sound-reproducing systems characterized in that: A: for each speaker hpj, at least one sound signal S is sent on the speaker hpj, for each microphone mi, a piece of information hpjmi is retrieved, this piece of information characterizing the sound-reproducing system comprising the speaker hpj and the microphone mi, B: a reference matrix Qr is saved, this reference matrix being constituted by all the pieces of reference information hpjmi obtained following the sending of the sound signal S, C: as soon as a comparison is to be made, the step A is run with a sound signal S? to obtain current information on a matrix Q, D: the matrices Q and Qr are compared.

Abstract: A transducer device includes an acoustic transducer, a parameter extractor and a feedback circuit. The parameter extractor is configured to extract an operating parameter from the acoustic transducer. The feedback circuit is configured to generate a correction signal based on a difference between the extracted operating parameter and a corresponding reference parameter. The correction signal is applied to adjust the operating parameter of the acoustic transducer.

Abstract: A loudspeaker system has a primary driver and an active radiator sealed in an enclosure where the active radiator is adapted to vary its operating characteristics to tune the sound pressure level and resultant frequency response generated by a primary driver. The primary driver and the active radiator share the same acoustic volume of the enclosure, i.e., the primary driver and the active radiator share a common acoustic compliance of the internal enclosure volume. The primary driver has electromagnetic components designed to oscillate a flexible cone or diaphragm along the longitudinal axis of the primary driver. The active radiator has electromagnetic components adapted to couple to a number of electrical configuration settings. Each electrical configuration setting may affect the operating characteristics of the diaphragm of the active radiator and is reflected back electro-acoustically, through the shared volume, to the primary driver.

Abstract: A speaker may include a center pole, a voice coil bobbin having a nonmetallic pipe body, and a first and a second electrodes which are provided on an inner peripheral face of the pipe body for detecting an electrostatic capacity. The first and the second electrodes are disposed so as to be separated from each other with a predetermined space along an axial direction of the center pole. Further, a stepped portion may be formed in the side circumferential face of the center pole such that, when the voice coil bobbin is operated, a first facing area of the first electrode to the side circumferential face defined by the stepped portion of the center pole increases while a second facing area of the second electrode to the side circumferential face defined by the stepped portion of the center pole decreases by a same amount as an increased amount of the first facing area.

Abstract: The system comprises a loudspeaker simulation unit for simulating the transmission behavior of the loudspeaker and comprises a room simulation unit, which is connected in outgoing circuit to the loudspeaker simulation unit and which is provided for simulating the transmission behavior of a given monitoring room. The room simulation unit is followed by a presentation unit, which generates an acoustic signal that corresponds to the auditory impression of the loudspeaker in the monitoring room, and/or is followed by an evaluation unit that evaluates the signal, which is provided by the room simulation unit, with regard to at least one psychoacoustic measured quantity, and the evaluation unit outputs a corresponding measurement signal. This measurement signal corresponds to a measurement signal that occurs inside the monitoring room during the presentation of the input signals.

Abstract: A method of processing a sound signal in an audio amplification device using frequency transposition, the method including the steps of: (a) receiving (11) an input sound signal, (b) determining (19) gains for amplifying the input sound signal at a plurality of input frequencies, (c) transposing (20) one or more of the input frequencies of the amplified sound signal to generate one or more output signals at transposed frequencies, (d) determining (31-33) the presence of an undesired feedback signal component resulting from the amplification and frequency transposition of the input sound signal at the input frequencies, and (e) correcting (34) the output signal at each of the transposed frequencies to compensate for the presence of the undesired feedback signal component.

Abstract: A system may include a processor and memory. The processor may be configured to perform calibration measurements on the speaker even when the speaker is being used to conduct a live conversation. The processor may be configured to: provide a live output signal for transmission from a speaker; receive an input signal corresponding to the output signal; compute a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of the input signal and a spectrum of the output signal; subtract the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; perform an iterative search for current parameters of a speaker model using the input signal spectrum, the output signal spectrum and the speaker-related sensitivity; and update averages of the speaker model parameters using the current parameter values. The parameter averages may be used to perform echo cancellation.

Abstract: A method and apparatus for detecting a singing frequency in a signal processing system using two neural-networks is disclosed. The first one (a hit neural network) monitors the maximum spectral peak FFT bin as it changes with time. The second one (change neural network) monitors the monotonic increasing behavior. The inputs to the neural-networks are the maximum spectral magnitude bin and its rate of change in time. The output is an indication whether howling is likely to occur and the corresponding singing frequency. Once the singing frequency is identified, it can be suppressed using any one of many available techniques such as notch filters. Several improvements of the base method or apparatus are also disclosed, where additional neural networks are used to detect more than one singing frequency.

Abstract: An acoustic apparatus includes an obtaining section configured to obtain impulse response data between at least one speaker and a microphone; a computation section configured to compute step response data by integrating the impulse response data obtained by the obtaining section; and a determination section configured to determine a connection polarity of the speaker in accordance with the size relationship of areas of a region on the positive side and a region on the negative side of the step response data in a determination segment of a predetermined time width in which a rise point of the step response is a starting point.

Abstract: Provided is a method for canceling background noise of a sound source other than a target direction sound source in order to realize highly accurate speech recognition, and a system using the same. In terms of directional characteristics of a microphone array, due to a capability of approximating a power distribution of each angle of each of possible various sound source directions by use of a sum of coefficient multiples of a base form angle power distribution of a target sound source measured beforehand by base form angle by using a base form sound, and power distribution of a non-directional background sound by base form, only a component of the target sound source direction is extracted at a noise suppression part. In addition, when the target sound source direction is unknown, at a sound source localization part, a distribution for minimizing the approximate residual is selected from base form angle power distributions of various sound source directions to assume a target sound source direction.

Abstract: A system such as a speakerphone may include a processor, memory, a speaker and a microphone. The processor may be configured (via program instructions stored in the memory) to calibrate the speaker by: outputting a stimulus signal; receiving an input signal corresponding to the stimulus signal; computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of the input signal and a spectrum of the output signal; subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; performing an iterative search for current parameters of a speaker model using the input signal spectrum, the stimulus signal spectrum and the speaker-related sensitivity; and updating averages of the speaker model parameters using the current parameter values. The stimulus signal may be transmitted during periods of silence in the external environment. The parameter averages may be used to perform echo cancellation.

Abstract: A broadside small array microphone beamforming apparatus comprises first and second omni-directional microphones, a microphone calibration unit, and a directional microphone forming unit. The first and second omni-directional microphones respectively convert voice from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are respectively arranged at three points of a triangle. The microphone calibration unit receives the first and second signals and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to generate a first directional microphone signal with a bidirectional polar pattern. The adaptive channel decoupling unit receives the first calibration signal and the first directional microphone signal to generate a first main channel signal and a first reference channel signal for noise detection.

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

Abstract: A transducer (1) for producing sound in response to an electrical signal comprises an actuator (2) with a magnet (4) and a coil (5), and a vibration surface (3), for example a loudspeaker cone. The actuator and the vibration surface are mechanically coupled. The transducer (1) is designed to operate at substantially its resonance frequency (f0). This results in a very high transducer efficiency, which is particularly relevant for rendering low audio frequencies.

Abstract: It is known to make the performance of a loudspeaker “environment adaptive” in controlling a filter unit based on a measurement of the velocity/acceleration of the loudspeaker diaphragm and the associated sound pressure in front of the diaphragm, by means of an accelerometer and a microphone, respectively, thereby determining the radiation resistance of the diaphragm. The two sensors have to exhibit a constant transfer function throughout the life time of the loudspeaker, which make them very expensive. With the invention it has been found that the accelerometer can be replaced by another microphone held in a small distance from the diaphragm, and this conditions the possibility of using the same microphone for both measurements, e.g. simply by physically moving the microphone from one position to another. It will then no longer be required to use long-time stable sensors, whereby the price of the sensor equipment can be reduced dramatically. Also alternative arrangements are disclosed.

Abstract: A system for canceling acoustic feedback includes an input for receiving a digital audio signal and a processor configured to detect acoustic feedback signal in the digital audio signal and to determine the frequency of the feedback signal The system also includes a plurality of bandpass filters for attenuating the feedback signal. The processor is further configured to: select a bandpass filter from among the plurality of bandpass filters. The selected bandpass filter comprises a response characteristic that attenuates parts of the signal at the frequency of acoustic feedback signal.

Abstract: The present invention is directed to a method and system for automatic calibration of an acoustic system. The acoustic system may include a source A/V device, calibration computing device, and multiple rendering devices. The calibration system may include a calibration component attached to each rendering device and a source calibration module. The calibration component on each rendering device includes a microphone. The source calibration module includes distance and optional angle calculation tools for automatically determining a distance between the rendering device and a specified reference point upon return of the test signal from the calibration component.

Abstract: An apparatus for suppressing feedback in an environment where a microphone and a loudspeaker are located, comprises a means for embedding a test signal into a loudspeaker signal, a microphone signal or a modified microphone signal, preferably by using a psychoacoustic masking threshold by using a pseudo-noise test signal, a means for determining a characteristic of a transmission channel in the environment between the loudspeaker and the microphone by using the embedded test signal and the microphone signal, a filter for filtering the loudspeaker signal to obtain a filtered loudspeaker signal, wherein the filter is adaptable to be adapted with regard to its filter characteristic to the characteristic of the transmission channel by the means for determining, as well as a means for subtracting the filtered loudspeaker signal from the microphone signal to obtain the modified microphone signal, in which the feedback is reduced due to the loudspeaker signal.

Abstract: The method comprises generating an acoustical volume velocity Q in the listening position, measuring a response quantity p, such as sound or vibration, at a suspected source position resulting from the volume velocity Q, and determining the acoustical transfer impedance Zt as the response quantity p divided by the acoustical volume velocity Q, Zt=p/Q. According to the invention the acoustical volume velocity Q is generated using a simulator (10) simulating acoustic properties of at least a head of a human being, the simulator comprising a simulated human ear (14, 15) with an orifice in the simulated head and a sound source (30) for outputting the acoustical volume velocity Q through the orifice. The output volume velocity Q from the orifice of an ear is estimated from measurements with two microphones inside the corresponding ear canal.

Abstract: A signal processing system improves signal quality by accurately locating and eliminating a feedback signal in an input signal, such as an audio signal. The signal processing system interpolates between frequency sample points to obtain a more accurate identification of a feedback signal frequency. A less intrusive filter reduces or eliminates the identified frequency signal frequency without excessive adverse effects on adjacent frequencies in the input signal.

Abstract: An electro acoustic system built-in test and calibration method utilizes a built-in self-test module to send a test signal through a first circuit device to an audio transmitter, causing the audio transmitter to output a test signal, for enabling the test signal to be received by a audio receiver and then processed by a connected circuit device and converted into a feedback digital signal to the self-test module for comparing the linearity relative to the originally provided test signal so that the parameter values and conformity of circuit devices can be optimized subject to comparison result. The test and adjustment procedure is recycled for other parameter items, and a warning signal is produced when proper adjustment cannot be done.

Abstract: In one embodiment, a display device includes a movably suspended diffuser screen, and a voice-coil mechanism coupled to the diffuser screen. A signal-processing device, such as a microprocessor, is coupled to a driver circuit to produce a drive signal at an output terminal for the voice coil in response to a voltage sensed across the voice coil. The driver circuit includes shutdown control to drive its output terminal to a high-impedance state to accommodate sensing voltage across the voice coil. Thus, the signal for the voice coil is produced by the signal-processing device in a closed-loop feedback arrangement without the need for separate position-sensing elements. In a preferred arrangement, a second voice coil is coupled to the diffuser screen and to the signal-processing circuitry to produce a second signal for the second voice coil to accommodate generating a circular motion for the diffuser screen without stationary points.

Abstract: An input sound processor compares power at each frequency component of an input sound with a reference value, and sets multiplication points indicating frequency components at which the total power of the input sound is to be determined. A product-sum operation is performed at the multiplication points on the power at each frequency component and the square amplitude of each filter coefficient indicating the transfer characteristic from a loudspeaker to a microphone to estimate the total power of the input sound at the position of the microphone.

Abstract: A power amplifier to provide a compensated output voltage to a load through a series-connected impedance. The power amplifier includes an inner positive current feedback loop that is capable of sensing changes in the resistance of the load, and which adjusts the effective impedance of the series-connected impedance seen by the load to reduce current induced changes in the level of the compensated output voltage provided to the load due to the presence of the series connected impedance.

Abstract: A speaker feedback system has a linear current feedback derived from a current sensing resistor. A linear motional feedback from the active speaker driver is derived from a sensing coil mechanically coupled with the driver coil of the speaker, the driver coil receiving electrical energy from the power amplifier. A motional signal correction circuit extracts the motional signal from the sensing coil and feeds to input B of the multiplier. A current filter circuit filters out higher-frequency signals from the current feedback signal and then feeds to input A of the multiplier. A multiplier performs the multiplication function of its two inputs and produces an output. Post-multiplier equalization circuitry that control the feedback gain of the nonlinear feedback system such that it can effectively compensate for the effect of flux modulation over the targeted frequency range. A feedback network feeds the output from post multiplier equalization to the power amplifier.

Abstract: An automatic sound field correcting device executes a signal process to the plurality of audio signals on respective correspondent signal transmission paths, and outputs them to a plurality of correspondent speakers to correct sound characteristics on the respective signal transmission paths. Namely, a measurement signal is supplied to each signal transmission path, and a measurement sound corresponding to it is outputted from the speaker to a sound space. The outputted measurement sound is detected as a detecting signal. The frequency characteristic of the audio signal on each signal transmission path is corrected by an equalizer, and a gain value of the equalizer is determined by a correction amount determining unit. A frequency characteristics correction is performed predetermined times. At a first correction, the correction amount determining unit determines the correction amount by performing a frequency analysis, based on the detecting signal, i.e.

Abstract: A signal delay time measurement device outputs, to a sound space, a measurement signal sound corresponding to a measurement signal such as a pulse signal, and obtains a response signal indicating a response thereof. By comparing the response signal with a predetermined threshold, the signal delay time measurement device measures a signal delay time in the sound space. The signal delay time in the above-mentioned sound space includes a delay time other than the delay time caused by a transmission of a signal sound to the sound space, and the response signal cannot theoretically reach the signal delay amount calculating unit during the delay time. Therefore, the delay time calculating unit does not perform the comparison in a no-response period in which the response signal has not reached the delay time calculating unit yet. Thereby, it can be prevented that the signal delay time is erroneously calculated by an effect of a background noise during the no-response period.

Abstract: A transducer senses sounds produced by a talker or other source and measures acceleration of air. Enhancement of acceleration is accompanied by reduction of the portion of the sound energy that escapes from the regions around the transducer. The result is a high sensitivity transducer, with increased privacy for use in communication systems, especially cell phones and in a multi-person environment. A pressure sensor array with a weighted output is sensitive to sound from a source talker only, and not to acoustic background noise, and not to a local loudspeaker. The weighted signal is a source sum pressure signal. The array produces a signal (using a different weighting) that corresponds to an estimate of a derivative of pressure. The derivative signal is proportional to the volume velocity fluctuations produced by the source. This signal is enhanced, rather than reduced. A local loudspeaker is driven to make the source sum pressure signal as small as desired.

Abstract: To improve the reproduction of audio signals, the signal components of a selected audio frequency range (1) of an audio signal are concentrated in a narrower audio frequency range (II). This is achieved by detecting first signal components in the first audio frequency range (I), generating second signal components in the second audio frequency range (II), and controlling the amplitude of the second signal components in response to the amplitude of the first signal components. As a result, dedicated transducers may be used which are particularly efficient in the narrower frequency range. The original frequency range (I) may contain the lower frequency signal components (bass components) of the audio signal.

Abstract: Embodiments of the present invention provide methods and devices for controlling a command signal applied to a load. In embodiments of the invention, current through and voltage across a load are determined and the values of both are used to generate a hybrid control signal. For example, in some embodiments the hybrid control signal is generated by taking a weighted summation of the current and voltage control signals. In other embodiments, a percentage of the difference between the current and voltage control signals is added to one of the current or voltage control signals to generate the hybrid control signal. In one embodiment, a potentiometer is used to generate the hybrid control signal.

Abstract: This invention provides a compensation system capable of compensating for power loss due to the power compression effects of the voice coil as the temperature of the voice coil increases. To compensate for the power compression effect, the invention predicts or estimates the temperature of the voice coil using a thermal-model, and adjusts the estimated temperature according to the cooling effect as the voice coil moves back and forth in the air gap. The thermal-model may be an equivalent electrical circuit that models the thermal circuit of a loudspeaker. With the input signal equating to the voltage delivered to the loudspeaker, the thermal-model estimates a temperature of the voice coil. The estimated temperature is then used to modify equalization parameters.

Abstract: A method of limiting the power applied to a loudspeaker is disclosed, in which both the voltage and current applied to the loudspeaker are measured, and instantaneous power is directly calculated and used to control the level of the input signal that drives the amplifier powering the loudspeaker. When the power applied to the loudspeaker exceeds a prescribed threshold, the input level to the power amplifier is reduced until the measured power falls below the threshold. Also disclosed is a method for indirectly determining the voice coil temperature from the loudspeaker's voltage and current and reducing power to the loudspeaker when the temperature exceeds a prescribed threshold. The power level is actively controlled to prevent damage to the loudspeaker and to minimize audibly objectionable artifacts.

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: A method and an apparatus for compensating for nonlinear distortion are provided to divide audio signals reproduced in a nonlinear speaker system into linear and nonlinear components in a time domain and a frequency domain, and then generate inversely-corrected signals by means of an inverse filtering scheme, so that it is possible to further consider a variety of nonlinear distortion characteristics such as viscous damping and structural damping which have not been reflected in the conventional lumped parameter method, and thus to obtain better sound quality.

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: Neural networks provide efficient, robust and precise filtering techniques for compensating linear and non-linear distortion of an audio transducer such as a speaker, amplified broadcast antenna or perhaps a microphone. These techniques include both a method of characterizing the audio transducer to compute the inverse transfer functions and a method of implementing those inverse transfer functions for reproduction. The inverse transfer functions are preferably extracted using time domain calculations such as provided by linear and non-linear neural networks, which more accurately represent the properties of audio signals and the audio transducer than conventional frequency domain or modeling based approaches. Although the preferred approach is to compensate for both linear and non-linear distortion, the neural network filtering techniques may be applied independently.

Abstract: A system (200) and method (400) for near-end detection of voice (107) in speakerphone mode is provided. The method can include determining (402) a convergence of an adaptive filter (220), determining (404) a dissimilarity between an autocorrelation (311) of an echo estimate (244) and an autocorrelation (312) of a microphone signal (243) if the adaptive filter has converged, computing (406) a weighting factor (279) based on the dissimilarity, applying the weighting factor to a voice activity level (281) to produce a weighted voice activity level (283), comparing (410) the weighted voice activity level to a constant threshold, and performing (412) a muting operation in accordance with the comparing for providing half-duplex communication.