Abstract: A nonlinear control system includes a controller, a model updater, and a model. The controller is configured to accept one or more input signals, and one or more updates generated by the model updater to produce one or more control signals. The system is configured to drive one or more transducers with the control signals to produce a rendered audio stream therefrom. The model updater is configured to analysis one or more portions of the audio stream and to update one or more aspects of the controller so as to alter performance of the transducer.

Abstract: There is provided a system and method for playing a venue-specific object-based audio in a venue, the system comprising a memory, and a processor configured to receive an object-based audio including a plurality of audio components and create a venue-specific object-based audio based on a modification metadata by adjusting a level of at least one of the plurality of audio components of the object-based audio, the processor executing the object-based audio rendering software to render the venue-specific object-based audio in the venue.

Abstract: Example techniques may involve performing aspects of a spectral calibration using an applied spatial calibration. An example implementation may include receiving data representing spatial filters that correspond to respective playback configurations. The implementation may also involve causing the audio drivers to output calibration audio that is divided into a repeating set of frames, the set of frames including a respective frame for each playback configuration. Causing the audio drivers to output the calibration audio may involve causing an audio stage to apply, during each frame, the spatial filter corresponding to the respective playback configuration. The implementation may also include receiving data representing spectral filters that correspond to respective playback configurations, the spectral filters based on the calibration audio output by the audio drivers.

Abstract: A corrector is configured to transform audio signals to compensate for unwanted distortion characteristics of a loudspeaker. A tuning filter is configured to transform audio signals to incorporate desired distortion characteristics associated with a target loudspeaker. By chaining together the tuning filter and the corrector, an audio signal can be modified so that the loudspeaker, when outputting the audio signal, has response characteristics of the target loudspeaker.

Abstract: A special sound producing device which is loaded in a vehicle, and provides information to a vehicle crew by a special sound, includes a plurality of speakers which become the power sources of the special sound, and a control unit which generates frequency signals to make the special sound output from the plurality of speakers respectively, wherein when one information to be transmitted to the vehicle crew is transmitted by the special sound from the plurality of speakers, the control unit makes the frequencies of the frequency signals, the sound pressures of the special sound or the output timings to the speakers of the frequency signals to be different among the plurality of speakers.

Abstract: An optimised method for driving a loudspeaker is used for protecting the loudspeaker from damage due to excessive excursion or from overheating. The playback power of an incoming audio data stream is compared with a feedback power derived from the loudspeaker actuator and the comparison is used to adjust the frequency response of the loudspeaker, across individual sub bands.

Abstract: In one embodiment, a pre-correcting engine generates a pre-corrected signal that may be used to mitigate air path distortions in sounds generated by loudspeaker systems that include loudspeakers coupled to waveguides. In operation, the pre-correcting engine virtually propagates a target signal in a reverse direction from the output of the waveguide to the output of a driver. Notably, the pre-correcting engine implements a propagation distortion compensation model that, when applied to the target signal, corrects for the effects of air path distortions on the target signal. The pre-correcting engine then transmits the pre-corrected signal to the driver. As the driver drives the waveguide based on the pre-corrected signal, effects of the pre-corrected distortions negate effects of air path distortions and the loudspeaker system generates high-fidelity sounds.

Abstract: Problem to be Solved To effectively operate a microphone array provided with a sound source position estimating function. Solution A microphone control system of the present invention is realized by the use of a tablet-type computer, for example. On the display of the control system, a two-dimensional simulation diagram 302, for example, simulating the space in which the microphone array is disposed. An array mark 304 simulating the exterior shape of the microphone array is displayed on the simulation diagram 302. Further, a sound source mark 316 indicating the position of the sound source estimated by the microphone array is displayed on the simulation diagram 302. Thus, an operator can intuitively find the position of the sound source in the space in which the microphone array is disposed, by referring to the simulation diagram 302, the array mark 304 and the sound source mark 316 displayed on the simulation diagram. Thus, the usage of the microphone array is improved.

Abstract: This application relates to methods and apparatus for thermal protection of a loudspeaker (102). A controller (106) is configured to generate a control signal (Gmod) for modulating the gain of a signal processing chain (101) that drives the loudspeaker. The controller is configured such that the control signal is generated as a function of each of: an indication of voice coil temperature of the loudspeaker (TVC); an indication of power dissipation in the voice coil of the loudspeaker (PVC); and an indication of a reservoir temperature (Tres) of a thermal reservoir for heat flow from the voice coil. The reservoir temperature may be an ambient temperature of the environment or a temperature of a component of the loudspeaker or apparatus that acts as a heat sink.

Abstract: This accessory is apparatus capable of implementing a method that can affect one or more features of a portable electronic device when brought into a vehicle having a radio with a right and left speaker. The accessory has a sender for initiating transmission of an RF test signal at the portable electronic device. The RF test signal is arranged to cause the radio to produce a distinct pair of audio responses from the right and left speaker. The accessory has a discriminator for determining for the right and left speaker the corresponding return delay between transmission of the RF test signal and arrival of the distinct pair of audio responses at the portable electronic device.

Abstract: A system for receiving multiple conversations or messages and for playing the multiple conversations or messages with a mobile device and wireless earpieces. The system may determine various presentation parameters based on various characteristics of the received messages and may play the messages such that audio appears at distinguishing locations around the user. The system may change how messages and/or conversations are played in response to recognize a change in the focus of the user based on detected user inputs, such as body movement gestures.

Abstract: A control circuit configured to control a speaker is disclosed. The control circuit is configured to drive the speaker at a predetermined frequency that may correspond to a resonance frequency. The control circuit comprises a speaker driver in communication with a controller. The controller is operable to control a drive frequency of the speaker by monitoring at least a sample of each of a plurality of wavelengths of a current draw of the speaker in the form of a voltage signal. Based on the voltage signal, the controller is operable to identify a voltage differential and adjust the drive frequency of the speaker in response to the voltage differential to maintain the predetermined frequency.

Abstract: The present application discloses an audio mixing method and system for audio mixing on original sound signals. The method includes: arranging a plurality of loudspeaker boxes according to predetermined positions to form a predetermined acoustic space, the predetermined acoustic space including a plurality of predetermined acoustic positions; and arranging, at predetermined acoustic positions in the predetermined acoustic space, sound track elements of each sound track among one or more sound tracks based on a predetermined rule. The present application provides an audio mixing method and system that implement subtle acoustic effects and provide better user experience.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

Abstract: Examples described herein involve calibrating a playback device for a playback environment based on audio signals detected by a microphone of a network device as the network device moves about the playback environment. While the playback device is playing a first audio signal and the network device is moving within the playback environment from a first physical location to a second physical location, the network device may detect by a microphone of the network device, a second audio signal. The network device may then identify an audio processing algorithm based on data indicating the second audio signal, and transmit to the playback device, data indicating the identified audio processing algorithm. Similar functions may also be performed by the playback device being calibrated or a computing device, such as a server to coordinate calibration of the playback device.

Abstract: The volume of a sound generated from a source is monitored. Contextual data is collected from sensors. Based on the collected contextual data listeners of the sound are identified. The identified listeners are grouped into listeners that are bothered by the sound and listeners that are unbothered by the sound. An acceptable range for the volume of the sound is determined based on the bothered listeners and the unbothered listeners.

Abstract: A close-type speaker leak test system includes a first chamber, a second chamber, a third chamber, a fourth chamber, a vacuum generator, and a differential pressure gauge. Firstly, a negative pressure value of the first chamber and the second chamber is generated by the vacuum generator. A standard close-type speaker is placed within the third chamber, and the first chamber and the third chamber are in communication with each other. An under-test close-type speaker is placed within a fourth chamber, and the second chamber and the fourth chamber are in communication with each other. A pressure difference value between the third chamber and the fourth chamber is measured by the differential pressure gauge. If the pressure difference value is larger than a predetermined value, the under-test close-type speaker has the leak. Consequently, the misjudgment is reduced, and the testing efficiency is enhanced.

Abstract: A display device for a motor vehicle includes: a display for displaying vehicle information; a computing device receiving vehicle data via a data bus, evaluating the received vehicle data, and causing an acoustic warning as a function of the evaluation of the vehicle data; and a connection between the display device and a loudspeaker for outputting of the acoustic warning, the connection to the loudspeaker being connected to an input of an evaluation device in such a way that, depending on a signal present at the connection of the loudspeaker, a malfunctioning or a functioning of the acoustic output is determined, and an optical warning display is provided for outputting an optical warning in the case of a malfunction of the acoustic output.

Abstract: An audio output device is described comprising a speaker; an audio output circuit configured to receive a first audio signal and configured to supply the first audio signal and a second audio signal to the speaker, wherein the second audio signal comprises a higher frequency than the first audio signal; a microphone configured to receive an acoustic signal from the speaker in response to the first audio signal and the second audio signal and to convert the acoustic signal into a received audio signal and a determiner configured to determine a phase of a frequency component of the received audio signal corresponding to the second audio signal and to determine an excursion of the speaker by the first audio signal based on the phase.

Abstract: The present technology relates to a sound processing apparatus and method, and a program for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: In one embodiment of the present invention, a loudspeaker parameter estimation subsystem efficiently and accurately estimates parameter values for a lumped parameter model (LPM) of a loudspeaker. In operation, the loudspeaker parameter estimation subsystem trains a neural network model based on responses generated via the lumped parameter model and the corresponding sets of parameter values. Subsequently, based on the relationship between the measured output response of a loudspeaker to an input stimulus, the loudspeaker parameter estimation subsystem estimates parameter values for the LPM of the loudspeaker. Advantageously, by sagaciously estimating parameter values for the LPM of loudspeakers, these NN-based techniques enable designers to leverage the LPM to reliably improve the design of loudspeakers, perform nonlinear correction of loudspeakers, and the like.

Abstract: Methods for modeling a loudspeaker having a passive radiator include applying a stimulus signal to a speaker within the cabinet, wherein the stimulus is applied over a frequency range. The sound pressure level (SPL) in the cabinet is measured as a function of frequency during application of the stimulus signal. At least one coefficient based on the measured SPL is derived, wherein at least one passive radiator parameter is a function of the at least one coefficient.

Abstract: Provided is an information processing system including a recognizing unit configured to recognize a first target and a second target on the basis of signals detected by a plurality of sensors arranged around a specific user, an identifying unit configured to identify the first target and the second target recognized by the recognizing unit, an estimating unit configured to estimate a position of the specific user in accordance with the a signal detected by any one of the plurality of sensors, and a signal processing unit configured to process each of signals acquired from sensors around the first target and the second target identified by the identifying unit in a manner that, when being output from a plurality of actuators arranged around the specific user, the signals are localized near the position of the specific user estimated by the estimating unit.

Abstract: Embodiments of systems and methods are described for inverting transformations of signals due to room acoustics. In some implementations, a transformation of a calibration signal from a particular location in a room may be determined. From this transformation, an inverse transformation may be determined and the inverse transformation may be applied to a speech signal received from a similar location.

Abstract: An audio system is described including an audio processor, an amplifier and a speaker and a distortion estimator. The distortion estimator calculates at least one of an expected response of the amplifier to an audio signal and an expected response of the loudspeaker to an audio signal. The distortion estimator is operable to generate a distortion prediction signal determined by a difference between an expected non-linear response of at least one of the loudspeaker and the amplifier and an expected linear response of at least one of the loudspeaker and the amplifier. A controller coupled to the audio processor and a control input of the audio processor may vary the operating parameters of the audio processor depending on the estimated distortion.

Abstract: A method of monitoring speaker temperature for speaker protection starts by generating a low level inaudible noise signal and injecting the low level inaudible noise signal in an audio signal. The voice coil resistance estimate that estimates a resistance of a voice coil of a speaker is then computed. The voice coil resistance estimate changes while the speaker is being driven by the audio signal that includes the low level inaudible noise signal. The temperature estimate is then computed based on the voice coil resistance estimate. The level of the audio signal may be adjusted based on the temperature estimate. Other embodiments are also described.

Abstract: In at least one embodiment, an apparatus for performing an audio measurement is provided. The apparatus includes a controller that is programmed to generate an audio test signal based on a first frequency range that comprises a first noise spectrum, a second frequency range that comprises a second noise spectrum, and a third frequency range that comprises a third noise spectrum and to transmit the audio test signal to an audio device that generates an audio response signal responsive to the audio test signal. The controller is further programmed to receive the audio response signal and to determine audio related characteristics for the audio device in response to the audio response signal.

Abstract: An audio output apparatus and a method are provided, which includes calculating a temperature value of a voice coil of the audio output apparatus using a heat transfer model algorithm, in response to power being supplied to the voice coil; adjusting an output level of an audio signal by determining a gain value to adjust the output level of the audio signal according to the calculated voice coil temperature value, and outputting the audio signal with the adjusted output level.

Abstract: A sound reproduction device includes a modulator having an output terminal for outputting a modulated carrier wave signal obtained by modulating a carrier wave signal in a ultrasonic band with an audible sound signal, a super-directivity loudspeaker connected to the output terminal, a capacitor connected between a ultrasonic wave source and a ground, first and second current detectors for detecting currents flowing through the super-directivity loudspeaker and the capacitor, a high-pass filter for outputting a filtered signal obtained by eliminating a low-frequency band component of the current detected by the first current detector, and a differential amplifier unit for outputting a signal corresponding to a difference between the filtered signal and the current detected by the second current detector. The ultrasonic wave source is configured to output the carrier wave signal such that the signal output from the differential amplifier unit is constant.

Abstract: Systems and methods for calibrating a loudspeaker with a connection to a microphone located at a listening area in a room. The loudspeaker includes self-calibration functions to adjust speaker characteristics according to effects generated by operating the loudspeaker in the room. In one example, the microphone picks up a test signal generated by the loudspeaker and the loudspeaker uses the test signal to determine the loudspeaker frequency response. The frequency response is analyzed below a selected low frequency value for a room mode. The loudspeaker generates parameters for a digital filter to compensate for the room modes. In another example, the loudspeaker may be networked with other speakers to perform calibration functions on all of the loudspeakers in the network.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

Abstract: A method for optimizing the design and sound field control of a compact loud-speaker array, which includes a plurality of loudspeakers located on a closed loudspeaker surface and the control of the emitted sound field by the loudspeakers within a limited reproduction subspace, having the steps of capturing the sound field using a plurality of microphones and adjusting filter coefficients that modify the alimentation signals of the loudspeakers to minimize the difference between reproduced signals captured by the microphones and target signals describing a target sound field. A conical reproduction surface encloses a reproduction subspace is defined such that the apex of the conical reproduction surface is within the closed loudspeaker surface. Loud-speakers are positioned on a limited loudspeaker surface and the closed loudspeaker surface.

Abstract: Methods and apparatus are disclosed for subwoofer calibration. An example method includes determining a crossover frequency of a playback system, and the playback system includes the subwoofer and a playback device. The example method includes the subwoofer and the playback device configured to output a multimedia content in synchronization. The example method also includes outputting from the subwoofer and the playback device a series of tones near the crossover frequency. The example method also includes prompting a user to select a tone set from the series of tones, and includes automatically adjusting a phase of the subwoofer in relation to the playback device based on the user selection.

Abstract: The present invention relates in one aspect to a voice coil temperature protector for electrodynamic loudspeakers. The voice coil temperature protector comprises an audio signal input for receipt of an audio signal supplied by an audio signal source and a probe signal source for generation of a low-frequency probe signal. A signal combiner is configured to combine the audio signal with the low-frequency probe signal to provide a composite loudspeaker drive signal comprising an audio signal component and a probe signal component. The voice coil temperature protector comprises a current detector configured for detecting a level of a probe current component flowing through the voice coil in response to the composite loudspeaker drive signal and a current comparator which is configured to comparing the detected level of the probe current component with a predetermined probe current threshold.

Abstract: An electronic device for buzz reduction is described. The electronic device is to be used with a speaker driver that is built onto an enclosure and is to be driven by an audio signal which could cause the enclosure to produce buzz. The electronic device includes a filter that is to attenuate a frequency component of the audio signal before driving the speaker driver. The electronic device also includes a controller that is to configure the filter to attenuate the frequency component of the audio signal in response to determining that strength of the audio signal at the frequency component exceeds a threshold. Other embodiments are also described and claimed.

Abstract: An apparatus is disclosed for inputting digital data on the output channel(s) of an audio subsystem in an audio device, without interfering with normal operation of the audio subsystem. The described circuit includes a resistive element in parallel with the expected load device, such as a headphone or speaker. The resistive element receives a modulated digital signal from a data source or a switch, and the instantaneous current through the resistive element due to the modulated digital signal is reflected in a current feedback mechanism of the audio subsystem. Demodulation logic retrieves the digital signal from the current measured by the current feedback mechanism. A capacitor is provided to prevent the current in the resistive element from the digital signal from impacting the average DC current that the feedback mechanism uses to evaluate the load device.

Abstract: The present invention is directed towards a method for collecting information pertaining to an audio notification system comprising a plurality of audio devices and a database, wherein each of the plurality of audio devices comprises a loudspeaker and a microphone, the method comprising: sequentially for each of the plurality of audio devices: emitting, by means of the loudspeaker of a currently emitting audio device, an audible test sound; monitoring, by means of microphones of the audio devices in the system, for the audible test sound; and for each monitoring audio device, if the audible test sound is detected by the monitoring audio device, storing, in the database, information pertaining to the currently emitting audio device and information pertaining to the monitoring audio device detecting the audible test sound.

Abstract: A loudspeaker control system is disclosed. The loudspeaker control system includes a loudspeaker, a sensor for measuring a voltage and current and a processor. The processor is adapted to calculate an input-voltage-to-excursion transfer function over time from an admittance function, blocked electrical impedance and force factor, use the input-voltage-to-excursion transfer function over time to predict an excursion and use the excursion to control audio processing of the loudspeaker.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

Abstract: A method and device for detecting a force factor of a loudspeaker is provided. The method includes the steps of: providing the loudspeaker with a dynamic driving voltage signal; continuously measuring a current signal flowing through the loudspeaker; observing the current signal and if the current signal shows that the diaphragm excursion of the loudspeaker is greater than an displacement limit, decreasing the driving voltage signal until the current signal shows that the diaphragm excursion of the loudspeaker is less than or equal to the displacement limit; and substituting the current driving voltage signal, the current signal, the displacement limit, and an electrical impedance of the loudspeaker into a function so as to compute the force factor of the loudspeaker.

Abstract: A power management system coupled to a speaker module includes a monitor module for measuring a current signal and a voltage signal of the speaker module to obtain a real-time impedance information of the speaker module; a reception module for receiving a time-domain audio analogy signal to be transformed into a frequency-domain audio digital signal; a prediction module for generating a power prediction information according to an initial audio information, the real-time impedance information and the audio digital signal; a control module for generating a control signal according to the audio digital signal and a human hearing model information; and a power adjustment module for outputting an adjustment audio signal to the speaker module according to the power prediction information, the audio digital signal and the control signal, so as to perform a broadcast operation of the speaker module.

Abstract: An integrated circuit for processing audio and voice signals in a telephone is disclosed. The integrated circuit comprises an audio output port having a first and a second output terminal for connection to a first and a second input terminal, respectively, of a speaker. The integrated circuit further comprises an audio input port having a first and a second input terminal for connection to a first and a second output terminal of an audio input source. The integrated circuit also comprises at least one audio output path and one audio input path. Moreover, the integrated circuit comprises a test impedance, such as a resistor, (RT) for measuring an electrical impedance of the speaker.

Abstract: An apparatus for measuring a plurality of loudspeakers arranged at different positions includes a generator of a test signal for a loudspeaker; a microphone device configured for receiving a plurality of different sound signals in response to one or more loudspeaker signals emitted by one of the loudspeakers in response to the test signal; a controller for controlling emissions of the loudspeaker signals by the loudspeakers and for handling the different sound signals so that a set of sound signals recorded by the microphone device is associated with each loudspeaker in response to the test signal; and an evaluator for evaluating the set of sound signals for each loudspeaker to determine at least one loudspeaker characteristic for each loudspeaker and for indicating a loudspeaker state using the at least one loudspeaker characteristic. This scheme allows automatic, efficient and accurate measurement of loudspeakers arranged in a three-dimensional configuration.

Abstract: Techniques for improved audio localization for visual effects are described. In one embodiment, for example, an apparatus may comprise a processor circuit and an audio management module, and the audio management module may be operable by the processor circuit to determine a position of a user interface element in a presentation area, determine an audio effect corresponding to the user interface element, determine audio location information for the audio effect based on the position of the user interface element, the audio location information defining an apparent position for the audio effect, and generate audio playback information for the audio effect based on the audio location information. Other embodiments are described and claimed.

Abstract: A directional sound apparatus is described. The apparatus comprises a receiver operable to receive audio data associated with a video stream and to receive, over a network, ancillary audio data containing supplemental audio data associated with the video stream; a storage device operable to store a plurality of profiles, each profile being associated with a different user of the sound system and defining the audio to be provided to the user; a user location device defining a first positional location of a first user having a stored profile; an audio output operable to connect to a plurality of speakers; and an audio controller operable to control the audio output to the plurality of speakers to simultaneously provide the audio data and the supplemental audio data to the first user at the first positional location and said audio data to a second user located at a second, different, positional location.

Abstract: A method for crest-factor based gain modification comprises sampling audio inputs to determine an energy for each input. The inputs are amplified to produce output channels. An average energy is determined for each frame by adding a percentage of a last average energy to a residual percentage of the energy. The last average energy is equal to zero for a first sampling of the audio inputs. The residual percentage is equal to one hundred minus the percentage of the last average energy. The amplifier gain is iteratively reduced by a decrease-step for a fixed period until a number of exceeding-inputs is less than or equal to a maximum. The exceeding-inputs have an average energy that exceeds a maximum-energy. The amplifier gain is iteratively increased by an increase-step for another fixed period until a number of exceeding-inputs is greater than a maximum.

Abstract: A method of virtually tuning an audio system that incorporates an acoustic compensation system, where the audio system is adapted to play audio signals in a listening environment over one or more sound transducers. The acoustic compensation system has an audio sensor located at a sensor location in the listening environment. The transfer functions from each sound transducer to the audio sensor location are inherent. The method contemplates recording noise at the sensor location, and creating virtual transfer functions from each sound transducer to the sensor location based on the inherent transfer functions from each sound transducer to the sensor location. Audio signals are processed through the virtual sound transducer to sensor location transfer functions. A virtual sensor signal is created by combining the audio signals processed through the virtual sound transducer to sensor location transfer functions with the noise recorded at the sensor location.

Abstract: In accordance with these and other embodiments of the present disclosure, systems and methods may include a controller configured to be coupled to an audio speaker, wherein the controller receives an audio input signal, and based on a displacement transfer function associated with the audio speaker, processes the audio input signal to generate an output audio signal communicated to the audio speaker, wherein the displacement transfer function correlates an amplitude and a frequency of the audio input signal to an expected displacement of the audio speaker in response to the amplitude and the frequency of the audio input signal.

Abstract: A device includes an audio processor, a first speaker, and a second speaker. The audio processor is configured to receive an audio signal. The first speaker is operatively coupled to the audio processor and is configured to produce a first sound wave. The first sound wave is associated with the audio signal. The second speaker is operatively coupled to the audio processor and is configured to produce a second sound wave. The second sound wave is configured to at least partially cancel the first sound wave.

Abstract: An audio communications device has a handset in which a touch sensing ear piece region is coupled to an acoustic leakage analyzer. The acoustic leakage analyzer is to analyze signals from the touch sensing ear piece region and on that basis adjust an audio processing parameter. The latter configures an audio processor which generates an audio receiver input signal for the device. Other embodiments are also described and claimed.