Abstract: Example techniques relate to audio generation in a media playback system. Based on one or more first functions and first characteristics of an area, the system may generate first audio that includes a first audio signal and a second audio signal. The system provides the first audio signal to at least one first audio driver and the second audio signal to at least one second audio driver, thereby causing a first playback device and a second playback device to play back the first audio synchronously. The system receives second characteristics of the area and based on one or more second functions and the second characteristics, generates second audio comprising a third audio signal and a fourth audio signal. The system provides the third audio signal to the at least one first audio driver and the fourth audio signal to the at least one second audio driver.

Abstract: The present technology relates to a sound output control apparatus and a sound output control method that perform control to prevent a balance of a system from being disturbed when reproduction is performed using a plurality of wireless speakers. A sound output control apparatus includes a transmitter that transmits first setting information regarding a sound output of the sound output control apparatus, a receiver that receives second setting information regarding a sound output of another sound output control apparatus, and a controller that controls a sound pressure or a reproduction frequency range in which reproduction is performed, on the basis of the first setting information or the second setting information. The first setting information and the second setting information each include at least information regarding a temperature, information regarding an amplitude, information regarding a remaining capacity of a power supply, and/or information regarding the reproduction frequency range.

Abstract: A method for determining a displacement offset of a voice coil of a speaker, comprising: providing a drive signal comprised of a program content signal and a test signal to the voice coil of the speaker; determining at least one value representative of an impedance of the voice coil according to a voice coil voltage and a voice coil current, wherein the voice coil voltage and the voice coil current result from, at least, the test signal; determining whether the at least one value indicates that an offset displacement of the voice coil exceeds a predetermined offset distance; and reducing a magnitude of the drive signal provided to the voice coil if the at least one value indicates that the offset displacement of the voice coil exceeds the predetermined offset threshold.

Abstract: An active noise reduction device includes an impulsive signal generator that generates an impulsive signal, a signal output terminal that outputs the impulsive signal to a loudspeaker, and a signal detector that detects a rising signal and a falling signal from a microphone signal outputted from a microphone collecting sound. The device also includes a connection determiner that determines, based on respective peak values of the rising signal detected and the falling signal detected, whether or not the loudspeaker is connected to the signal output terminal. The device further includes a polarity determiner that determines a connection polarity between the loudspeaker and the signal output terminal based on respective timings of the rising signal and the falling signal with respect to the impulsive signal, when it is determined that the loudspeaker is connected to the signal output terminal.

Abstract: Systems, methods, and apparatus are provided for recording high output power levels of sound at low sound pressure levels. For example, an apparatus comprises an enclosure, a speaker disposed within the enclosure, a microphone disposed within the enclosure, and an evacuation port. The evacuation port is configured to connect to a system that reduces a pressure level within the enclosure to a level that is less than an ambient air pressure level outside the enclosure. The enclosure is sealed or otherwise configured to provide a sealed enclosure, to maintain the reduced air pressure within the enclosure. The speaker can be driven by an amplifier at high output power levels to generate a distorted sound of an amplified electric musical instrument for recording purposes, while the reduced pressure level within the enclosure serves to attenuate the sound pressure level and perceived loudness which emanates from the speaker.

Abstract: Acoustic transducer systems and methods of operating acoustic transducer systems are provided. The methods can involve: receiving an input audio signal; determining a position of a diaphragm; determining a correction factor, a motor force factor, a spring error factor, and a system spring factor based at least on the position of the diaphragm; determining a corrected voice coil current based at least on the input audio signal, the correction factor, the spring error factor, and a velocity of the diaphragm; and applying a corrected audio signal to a voice coil fixed to the diaphragm based at least on the corrected voice coil current, wherein the corrected audio signal corrects the input audio signal to compensate for non-linear characteristics of the acoustic transducer system.

Abstract: A directivity adjustment device that maintains a constant direct-to-reverberant ratio based on the detected location of a listener in relation to the speaker array is described. The directivity adjustment device may include a distance estimator, a directivity compensator, and an array processor. The distance estimator detects the distance between the speaker array and the listener. Based on this detected distance, the directivity compensator calculates a directivity index form a beam produced by the speaker array that maintains a predefined direct-to-reverberant sound energy ratio. The array processor receives the calculated directivity index and processes each channel of a piece of sound program content to produce a set of audio signals that drive one or more of the transducers in the speaker array to generate a beam pattern with the calculated directivity index.

Abstract: A media system and a method of using the media system to reproduce spatial audio based on head-to-torso orientation, are described. The method includes determining a head-to-source orientation and a head-to-torso orientation based on head orientation data generated by a head tracking device. Determining the head-to-torso orientation includes determining torso movements based on movements of the head. The torso can be determined to move when the head movements meet a head movement condition, such as a predetermined angle of movement or pattern of movement. A binaural audio filter that is based on a head-related transfer function corresponding to both the head-to-source orientation and the head-to-torso orientation is applied to an audio input signal to generate an audio output signal. The audio output signal is played to accurately recreate spatial audio having sounds emitted to the user by a sound source. Other aspects are also described and claimed.

Abstract: The disclosure includes a voice isolation system comprising an acoustic echo-cancelation subsystem configured to receive a plurality of input signals, subtract an interference component from the input signals, and provide a plurality of output signals. The system also includes an adaptive beamformer subsystem configured to receive the plurality of output signals from the acoustic echo-cancelation subsystem and compute a signal-to-noise ratio enhanced signal based on the received output signals. The system also includes a residual noise suppressor subsystem configured to attenuate at least one portion of the SNR enhanced signal received from the adaptive beamformer subsystem based on the at least one portion having an SNR below a predetermined SNR threshold. The system also includes an automatic gain control subsystem configured to process a signal outputted from the residual noise suppressor subsystem and transmit a resulting signal as an output signal.

Abstract: One embodiment provides a method comprising inputting into a first layer of a neural network at least one data sample of a current displacement of one or more moving components of a loudspeaker and at least one or more other data samples of one or more prior displacements of the one or more moving components. The method comprises generating in the first layer a data vector by applying a nonlinear activation function based on the at least one data sample of the current displacement and the at least one or more other data samples of the one or more prior displacements. The method comprises inputting into a second layer of the neural network the data vector and applying an affine transformation to the data vector. The method comprises outputting a prediction of a current voltage to apply to the loudspeaker based on the affine transformation applied to the data vector.

Abstract: A signal processing apparatus includes: an audio signal processing unit configured to perform wavefront synthesis processing for at least part of a plurality of sound source data; a first output unit configured to output N-channel audio signals output from the audio signal processing unit to a first speaker device; a mix processing unit configured to mix the N-channel audio signals output from the audio signal processing unit; and a second output unit configured to output an audio signal output from the mix processing unit to a second speaker device.

Abstract: Monitoring circuitry, comprising: a current monitoring unit operable to monitor a speaker current flowing through a speaker and generate a monitor signal indicative of that current; and a controller operable, based on a control signal, to control the current monitoring unit to control whether the monitor signal is generated and/or a property of the monitor signal.

Abstract: A management server controls, by use of a first database, a first audio client system that includes a first speaker and a second speaker. The first database includes a first standard characteristic impedance, which is obtained by application of statistical processing to a first characteristic impedance of the first speaker and a second characteristic impedance of the second speaker. The management server acquires a new characteristic impedance of the first speaker, compares the new first characteristic impedance with the first standard characteristic impedance, to determine a condition of the first speaker, based on a result of the comparison. The management server updates, based on the condition of the first speaker, the first standard characteristic impedance of the first database, using the new first characteristic impedance.

Abstract: Wireless-enabled loudspeaker includes a wooden capacitive touch user interface. The loudspeaker may comprise at least one electroacoustic transducer, a processor in communication with the at least one electroacoustic transducer, and a wooden exterior surface comprising a capacitive touch user interface that allows a user to control operation of the loudspeaker. The wooden exterior surface acts a dielectric for the capacitive touch user interface. The loudspeaker may comprise a wireless transceiver circuit for receiving and transmitting wireless communication signals via a wireless network. The wireless transceiver circuit may receive wirelessly audio content from streaming audio content servers that are connected to the Internet. The capacitive touch user interface comprises a plurality of user control icons etched in the wooden exterior surface, and a plurality of capacitive sense electrodes located under the wooden exterior surface.

Abstract: A dry audio signal may be received. An envelope of the dry audio signal may be compared to a threshold of the playback device to determine a dry excess. A filter may be applied to the dry audio signal to generate a wet audio signal. An envelope of the wet audio signal may be compared to the threshold to determine a wet excess. Based on the dry and wet excess, a dry gain may be determined for the dry audio signal and a wet gain may be determined for the wet audio signal. The wet audio signal and the dry audio signals may be mixed in proportion to respective gains. The mixed audio signal may avoid exceeding an operational limit associated with audio playback on the playback device.

Abstract: This disclosure describes a ceiling tile microphone that includes: a plurality of microphones coupled together as a microphone array used for beamforming, the plurality of microphones are positioned at predetermined locations; a single ceiling tile with an outer surface on the front side of the ceiling tile where the outer surface is acoustically transparent, the microphone array combines with the ceiling tile as a single unit, the ceiling tile being mountable in a drop ceiling in place of a ceiling tile included in the drop ceiling; where the ceiling tile microphone further includes beamforming, acoustic echo cancellation, and auto voice tracking; where the ceiling tile microphone is used in a drop ceiling mounting configuration; where the microphone array couples to the back side of the ceiling tile and all or part of the ceiling tile microphone is in the drop space of the drop ceiling.

Abstract: According to one embodiment, a method, computer system, and computer program product for operating a flexible device to maintain a threshold level of audio quality through various changes in the flexible device's bend profile is provided. The present invention may include identifying the bending profile of a flexible device comprising one or more audio devices; and responsive to identifying one or more of the audio devices as occluded audio devices based on the bending profile, operating the flexible device to change the position of one of the one or more occluded audio devices to raise an audio quality associated with the occluded audio device above the threshold level of desired audio quality.

Abstract: Systems and devices configured by an algorithm to prevent or limit a speaker over-excursion are disclosed. The disclosed algorithm is computationally efficient because it exploits a relationship between an audio signal and a speaker's excursion that exists at low-frequencies, below a self-resonance of a speaker. The disclosed algorithm combines the low-frequency excursion protection with a high-frequency, transient excursion protection. The combined approach allows the transient excursion protection to use a shorter delay than otherwise possible. The shorter delay allows for a compressor to apply attenuation to a transient audio signal before a momentum of the speaker, caused by the transient audio signal, grows too large to be controlled.

Abstract: System, method, and various embodiments for providing a speaker normalization system are described herein. An embodiment operates by identifying a plurality of speakers connected to an audio device, wherein the audio device includes an internal speaker for which first capability information is known. Second capability information of each of the plurality of speakers is retrieved. User input indicating a relative position of each speaker is received. An audio configuration is calculated based on the second capability information and the relative position of each of the plurality of speakers. The calculated audio configuration is provided to each of the plurality of speakers, wherein each of the plurality of speakers is configured to receive and audibly output audio from the audio device based on the audio configuration.

Abstract: An IP speaker system is disclosed that includes an IP speaker connected to one or more external analog speakers. The IP speaker comprises a CPU that provides an audio data stream for transmission to a first amplifier and a second amplifier. The first amplifier outputs a first analog audio signal at a first power level for driving an internal driver of the IP speaker, and the second amplifier outputs a second analog audio signal at a second power level for driving one or more external drivers of the one or more external analog speakers. The CPU supervises the one or more external analog speakers so as to detect any speaker failure, disconnection, short or other change in operational status. The IP speaker also comprises a power circuit that receives electrical power transmitted over a network cable as its input power and provides output power available for consumption by the IP speaker and the one or more external analog speakers.

Abstract: A memory stores a linear model predicting a preference rating for in-ear headphones. A processor is programmed to generate a headphone response curve defining a frequency response of a headphone, apply the linear model to the headphone response curve to determine a preference rating, and provide the preference rating to predict overall sound quality of the in-ear headphone without listening tests.

Abstract: Examples described herein involve validating motion of a microphone during calibration of a playback device. An example implementation involves a mobile device detecting, via one or more microphones, audio signals emitted from one or more playback devices as part of a calibration process. After the one or more playback devices emit the audio signals, the mobile device determines whether the detected audio signals indicate that sufficient horizontal translation of the mobile device occurred during the calibration process. When the detected audio signals indicate that insufficient horizontal translation occurred, the mobile device displays a prompt to move the mobile device more while the one or more playback devices emit one or more additional audio signals as part of the calibration process. When the detected audio signals indicate that sufficient horizontal translation occurred, the mobile device calibrates the one or more playback devices with a calibration based on the detected audio signals.

Abstract: A method of monitoring electrical loads is disclosed. In an embodiment the method includes generating a first voltage signal and a second voltage signal, the second voltage signal in quadrature to the first voltage signal, injecting one of the first voltage signal or the second voltage signal into a signal propagation path towards an electrical load, sensing a current signal flowing through the electrical load as a result of the one of the first voltage signal or the second voltage signal injected into the signal propagation path and processing the first voltage signal, the second voltage signal and the sensed current signal.

Abstract: Systems and methods for protecting a loudspeaker from excessive excursion include an audio source, an adaptive excursion protection filter, an audio clipper, an inverse excursion protection filter, an amplifier and a loudspeaker. The system performs operations including receiving an audio signal, applying an excursion protection filter, the excursion protection filter adapting in real-time to one or more speaker conditions, clipping the audio signal, applying an inverse excursion protection filter, and amplifying, using an amplification circuit, the audio signal for output to the speaker.

Abstract: A device to process an audio signal representing output sound includes one or more processors configured to generate, responsive to sensor data indicative of pressure detected at a housing of the device, output data based on a predicted effect of the pressure on an acoustic output of the device. The one or more processors are also configured, responsive to the output data, to adjust operation of an audio playback component that generates the acoustic output.

Abstract: A system, method, and computer-readable medium are disclosed for performing an audio output detection operation. The audio detection operation includes: identifying a reference audio output associated with a manufacturing environment; calibrating a threshold noise level of the manufacturing environment; monitoring the manufacturing environment for an audio output above a predefined noise threshold; comparing the audio output above the predefined noise threshold with the reference audio output; and, present information associated with the audio output above the predefined noise threshold upon detection of a match between the audio output above the predefined noise threshold and the reference audio output.

Abstract: The present technology relates to an HRTF measurement method, an HRTF measurement device, and a program which can more simply acquire a head-related transfer function of an individual. The HRTF measurement device displays an image indicating a target direction that a user is to face. In a case where a front direction of the user and the target direction are matched, the HRTF measurement device outputs a measurement sound from a speaker and measures an HRTF on the basis of a result of receiving the measurement sound through a microphone mounted on an ear of the user. The present technology can be applied to, for example, a device which measures a head-related transfer function and the like.

Abstract: Embodiments described herein generally relate to measuring and evaluating a test signal generated by a device under test (DUT). In particular, the test signal generated by the DUT may be compared to a reference signal and scored based on the comparison. For example, a method may include: capturing a test signal from a device under test; splicing the test signal into a plurality of test audio files based on a plurality of frequency bins; at each frequency bin, comparing each of the plurality of test audio files to a corresponding reference audio file from among a plurality of reference audio files, the plurality of reference audio files being associated with a reference signal; and calculating a performance score of the device under test based on the comparisons.

Abstract: A system for driving an audio speaker that has an impedance, the system comprising: a scaler having an analog input and a scaler output that is an attenuated analog input; an amplifier having an amplifier input coupled to the scaler output and an amplifier output; a current sensor having a sensor input coupled to the speaker and a sensor output, the current sensor configured to generate a sensor signal at the sensor output responsive to a current from the audio speaker; and a compensator circuit having a circuit input coupled to the sensor output and a circuit output coupled to the amplifier input, the compensator circuit configured to supply a compensation signal to the amplifier input by applying a transfer function to the sensor signal.

Abstract: An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

Abstract: Embodiments described herein generally relate to measuring and evaluating a test signal generated by a device under test (DUT). In particular, the test signal generated by the DUT may be compared to a reference signal and scored based on the comparison. For example, a method may include: capturing a test signal from a device under test; splicing the test signal into a plurality of test audio files based on a plurality of frequency bins; at each frequency bin, comparing each of the plurality of test audio files to a corresponding reference audio file from among a plurality of reference audio files, the plurality of reference audio files being associated with a reference signal; and calculating a performance score of the device under test based on the comparisons.

Abstract: A method and system for calibrating an integrated volume acceleration sensor of a loudspeaker, wherein the method includes driving the loudspeaker with a calibration signal and meanwhile generating a sensor output signal by the integrated volume acceleration sensor measuring a volume acceleration over time of a motion element fixed to a moving part of the loudspeaker and/or of the moving part while the loudspeaker is driven with the calibration signal as well as generating a reference output signal by a reference sensor measuring the volume acceleration over time of the motion element and/or of the moving part of the loudspeaker while the loudspeaker is driven with the calibration signal, and additionally includes calculating a calibration value for the integrated volume acceleration sensor based on a ratio of the sensor output signal and the reference output signal and based on a predetermined reference calibration value of the reference sensor.

Abstract: Examples associated with audio component adjusting are described. One example includes storing a degradation for an audio component embedded in a device. A characteristic of the audio component is compared to the degradation model for the audio component. When the characteristic of the audio component indicates the audio component has reached a certain point on the degradation model, an attribute of the audio component is adjusted.

Abstract: An audio output system for energizing a multiple voice coil transducer supplies at least two power output signals to the voice coils, a pilot tone generator for generating a pilot tone signal, and a power output circuit. The power output circuit generates power output signals from the pilot tone and the input signal so that the voice coils respond to the input signal with an in-phase electro-mechanical relationship and respond to the pilot tone with an out-of-phase (motion canceling) electro-mechanical relationship, reducing the effect of the pilot town on mechanical movement of the voice coil. A sensing circuit senses electrical signal values at terminals of the at least two voice coils, and a processing circuit detects a response of the output transducer to the pilot tone and determines at least one operating characteristic of the output transducer from the electrical signal values.

Abstract: A speaker includes a frame, a magnet, a diaphragm, a first coil, and a second coil assembly. The second coil assembly includes at least one second coil group, and each second coil group includes two second coils. The magnet and the diaphragm are mounted on the frame. The first coil and each second coil are coupled to the diaphragm. The first coil is configured to drive the diaphragm to vibrate. The second coil assembly is configured to, when the diaphragm vibrates and a difference between vibration displacements of two vibration regions in the diaphragm that are correspondingly coupled to two second coils in any second coil group exceeds a preset threshold, drive at least one of the two vibration regions to move to reduce the difference between the vibration displacements of the two vibration regions.

Abstract: A correction system and a correction method of signal measurement are provided. In the method, a transmitted signal and a received signal are divided into a plurality of transmitted signal groups and a plurality of received signal groups according to a time length, respectively. The received signal is related to a signal received after the transmitted signal is transmitted, and the transmitted signal is a periodic signal. A plurality of to-be-evaluated groups are selected from the received signal groups according to a correlation between the transmitted signal groups and the received signal groups. The correlation corresponds to a delay between the transmitted signal and the received signal. The signal energy of the received signal is determined according to the signal energy of the to-be-evaluated groups. Accordingly, the accuracy of signal measurement can be improved.

Abstract: A method (200) of operating a hearing aid system with a very low delay. The invention also provides a hearing aid system (100, 300, 400) adapted for carrying out such a method.

Abstract: A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is separately processed with an early part and a late reverberation of the room impulse response, and the processed early part of the audio signal and the reverberated signal are combined. A transition from the early part to the late reverberation in the room impulse response is reached when a correlation measure reaches a threshold, the threshold being set dependent on the correlation measure for a selected one of the early reflections in the early part of the room impulse response.

Abstract: An augmented reality environment allows interaction between virtual and real objects. Multiple microphone arrays of different physical sizes are used to acquire signals for spatial tracking of one or more sound sources within the environment. A first array with a larger size may be used to track an object beyond a threshold distance, while a second array having a size smaller than the first may be used to track the object up to the threshold distance. By selecting different sized arrays, accuracy of the spatial location is improved.

Abstract: An acoustic characteristic measurement device includes a speaker configured to output an acoustic signal in which a watermark signal is embedded; a microphone configured to perform sound collection with respect to the acoustic signal output from the speaker in a predetermined listening position; and an acoustic characteristic measurement unit configured to determine an acoustic characteristic of an acoustic space from the speaker to the listening position, on the basis of the watermark signal embedded in the acoustic signal which is subjected to the sound collection by the microphone.

Abstract: A system and method for detecting the occurrence of a gunshot are provided. The method may include receiving acoustic signals at a microphone indicative of a magnitude of the acoustic signal at a plurality of discrete frequencies, converting the acoustic signals to a digital signal and then employing a Goertzel algorithm based digital signal filter on the digital signal to produce Goertzel magnitudes at the plurality of discrete frequencies. The method subsequently determines if a maximum value of the digital signal is higher than a predefined threshold value. If the threshold value is exceeded, the method predicts if the Goertzel magnitudes at the plurality of discrete frequencies are indicative of a gunshot sound via gradient boosting; and, transmits a signal indicative of a gunshot occurrence after predicting if a possible gunshot event has occurred. A debouncer may also be utilized to suppress duplicative signals.

Abstract: The present invention discloses a signal processing device, a signal processing method and a speaker. The signal processing device comprises a multi-band dynamic range controller, wherein the multi-band dynamic range controller receives an audio signal and includes a first band splitting unit and a resonant band adjustment unit; the first band splitting unit is configured to split the audio signal into multiple bands and obtain at least one resonant band therefrom, which has a resonant frequency band signal in a resonant frequency range of the audio signal; and the resonant band adjustment unit is configured to adjust the resonant frequency band signal based at least on a resonant band dynamic range control gain and output an adjusted resonant frequency band signal for combination with other band signals into a compression output signal.

Abstract: A robust alarm system has an alarm controller adapted to input an alarm trigger and to generate at least one alarm drive signal in response. Alarm subsystems input the alarm drive signal and activate one or more of multiple alarms accordingly. A subsystem function signal provides feedback to the alarm controller as to alarm subsystem integrity. A malfunction indicator is output from the alarm controller in response to a failure within the alarm subsystems.

Abstract: A method (100) for calibrating a sound delivery system (1) having a processing assembly, a data communications assembly (9) coupled to the processing assembly, and at least one audio transducer (21a, 21b) mounted with at least one processor (11) of the processing assembly and responsive thereto for delivering sound to a user (3), the method including the steps of: transmitting from a remote user interface device (6) for the sound delivery system, a sequence of command codes for specifying predetermined characteristics of test sounds; receiving the command code sequence at the communications assembly of the sound delivery system; providing the command code sequence to the processing assembly of the sound delivery system; reproducing by a selected at least one audio transducer, the predetermined test sounds under control of said at least one processor according to the command code sequence; measuring with a reference SPL meter (70) proximate to the audio transducer, characteristics of test sounds reproduced by

Abstract: A method of multi-factor authentication includes receiving, by a remote hosting server from a terminal, a request from a user possessing a trusted device to access a remote service. The remote hosting server generates challenge chirp signal information and sends the challenge chirp signal information to the terminal and the device. Measurements are received of a room impulse response taken by each of the terminal and the trusted device using the chirp signal information. It is checked whether a location of the terminal is known based on a measurement of the room impulse response. The measurements of the room impulse response of the terminal and the trusted device are compared. A level of access to the remote service is granted to the user based on whether the location of the terminal is known and whether the trusted device is present at the location of the terminal.

Abstract: The disclosure relates to an audible indicator in combination with a drug delivery device having a resilient force member capable of resting in either of two states. In a biased state, the resilient force member is biased and stores energy and in a relaxed state, the resilient force member is relaxed. The resilient force member releases the stored energy when changing from the biased state into the relaxed state, thereby generating an audible signal, wherein the resilient force member is supported by a retaining element in the biased state in order to prevent transition into the relaxed state.

Abstract: An audio signal processing device and an audio signal adjusting method thereof. The audio signal processing device includes a gain determining circuit and an audio signal processing circuit. The gain determining circuit receives an analog audio signal. The gain determining circuit determines a gain based on a difference between a predefined threshold and an amplitude of the analog audio signal in response to detecting that a current sampling amplitude of the analog audio signal is greater than the predefined threshold. The audio signal processing circuit is coupled to the gain determining circuit and receives the analog audio signal and the gain. The audio signal processing circuit attenuates the analog audio signal according to the gain to generate an output audio signal.

Abstract: One embodiment provides a method for nonlinear control of a loudspeaker. The method comprises determining a target displacement of a diaphragm of a speaker driver of the loudspeaker based on a physical model of the loudspeaker and a source signal for reproduction via the speaker driver. The method further comprises determining, based on the target displacement, a corrected drive signal that compensates for action of reluctance force on the speaker driver during the reproduction of the source signal, and transmitting the corrected drive signal to a power amplifier connected to the loudspeaker. The power amplifier drives the speaker driver based on the corrected drive signal. Audio distortion generated by the speaker driver during the reproduction of the source signal is reduced based on the power amplifier output, via the corrected drive signal.

Abstract: A signal processing device includes a switching receptor, a storage, and a signal processor. The switching receptor receives switching of a speaker serving as a supply destination of a signal. The storage stores an optimal setting, which is obtained by measuring characteristics of the speaker selected by the switching, in association with the switching of the speaker. The signal processor reads out the optimal setting, which is associated with the switching received by the switching receptor, from the storage, and uses the optimal setting to process the signal to be supplied to the speaker.

Abstract: Wireless-enabled loudspeaker includes a wooden capacitive touch user interface. The loudspeaker may comprise at least one electroacoustic transducer, a processor in communication with the at least one electroacoustic transducer, and a wooden exterior surface comprising a capacitive touch user interface that allows a user to control operation of the loudspeaker. The wooden exterior surface acts a dielectric for the capacitive touch user interface. The loudspeaker may comprise a wireless transceiver circuit for receiving and transmitting wireless communication signals via a wireless network. The wireless transceiver circuit may receive wirelessly audio content from streaming audio content servers that are connected to the Internet. The capacitive touch user interface comprises a plurality of user control icons etched in the wooden exterior surface, and a plurality of capacitive sense electrodes located under the wooden exterior surface.