Abstract: A participant computing device connected to a teleconferencing session receives mixed audio data from a teleconference computing system orchestrating the session. The mixed audio data comprises a mix of audio captured by other devices connected to the session. The mixed audio and audio captured by the other devices prior to receipt of the mixed audio data are compared to determine that the participant computing device is co-located with at least one of the other devices. Information indicating that some of the other devices are featured devices that most actively contribute to the mixed audio is obtained. Co-location information for each of the featured devices is received. The co-location information is indicative of whether a featured device is co-located with the participant computing device. Based on the co-location information, a determination is made that one of the featured devices is co-located with the participant computing device.

Abstract: Disclosed is a wireless audio device, which includes at least one microphone, at least one speaker, at least one sensor, a processor, and a memory storing instructions, and the instructions that, when executed by the processor, cause the wireless audio device, while the wireless audio device outputs a sound for reducing an outside sound acquired through the at least one microphone through the at least one speaker, to identify a specified outside sound of the outside sound acquired through the at least one microphone, to output a notification sound for indicating that the specified outside sound is identified through the at least one speaker, to identify a motion of a user of the wireless audio device through the at least one sensor, in response to outputting the notification sound, and to stop the output of the sound for reducing the outside sound based on the identified motion.

Abstract: The present disclosure is directed to a device and method for detecting presence or absence of human speech. The device and method utilize a low-power accelerometer. The device and method generate an acceleration signal using the accelerometer, filter the acceleration signal with a band pass filter or a high pass filter, determine at least one calculation of the filtered acceleration signal, detect a presence or absence of a voice based on the at least one calculation, and output a detection signal that indicates the presence or absence of the voice. The device and method are well suited for portable audio devices, such as true wireless stereo headphones, that have a limited power supply.

Abstract: Embodiments include systems with active sound canceling properties, fenestration units with active sound canceling properties, retrofit units with active sound canceling properties and related methods. In an embodiment a system can include a sound cancellation device include a sensing element to detect vibration of a transparent pane and/or a sound input device configured to detect sound incident on the transparent pane, as well as a vibration generator configured to vibrate the transparent pane and a sound cancellation control module. The sound cancellation control module can evaluate the detected vibration of the transparent pane at two or more discrete frequency bands. The sound cancellation control module can cause the vibration generator to vibrate the transparent pane causing destructive interference with sound waves at the two or more discrete frequency bands. Other embodiments are also included herein.

Abstract: At least one exemplary embodiment is directed to a method of earphone manufacturing with an ear canal microphone, ambient sound microphone and a speaker.

Abstract: The invention discloses a MIMO different-factor full-form model-free control method with parameter self-tuning. In view of the limitations of the existing MIMO full-form model-free control method with the same-factor structure, namely, at time k, different control inputs in the control input vector can only use the same values of penalty factor and step-size factors, the invention proposes a MIMO full-form model-free control method with the different-factor structure, namely, at time k, different control inputs in the control input vector can use different values of penalty factors and/or step-size factors, which can solve control problems of strongly nonlinear MIMO systems with different characteristics between control channels widely existing in complex plants. Meanwhile, parameter self-tuning is proposed to effectively address the problem of time-consuming and cost-consuming when tuning the penalty factors and/or step-size factors.

Abstract: A bicycle shifting device calibration system, a bicycle seatpost calibration system and a bicycle suspension calibration system are provided that allow for easy setting of a shifting position or an actuation parameter. The bicycle shifting device calibration system includes an electronic controller and a memory. The electronic controller sets a plurality of shifting positions related to a shifting device based on vibration information that is output from a vibration detector in a case where a movable portion of the shifting device is continuously moved from a first position to a second position, which differs from the first position, in a state where the shifting device is coupled to a bicycle. The memory stores the plurality of shifting positions set by the electronic controller.

Abstract: A method for controlling a home appliance in consideration of a state of a surrounding environment is disclosed. A method for controlling a home appliance according to an exemplary embodiment of the present disclosure includes operating a home appliance using a first operation profile, identifying at least one of a behavior type of a user or a type of sound around the user, using Internet of Things (IoT) communication using 5G communication and artificial intelligence (AI), and changing an operation profile for operating a home appliance to a second operation profile, based on at least one of the identified behavior type of the user or type of sound around the user. According to the present disclosure, in accordance with the state of the surrounding environment, the operation noise can be reduced or the operation time of the home appliance can reduced without increasing the discomfort of the user, and therefore the discomfort of the user due to the operation noise can be relieved.

Abstract: The amount of far-field noise transmitted by a primary communication device in an open-plan office environment is reduced by defining an acoustic perimeter of reference microphones around the primary device. Reference microphones generate a reference audio input including far-field noise in the proximity of the primary device. The primary device generates a main audio input including the voice of the primary speaker as well as background noise. Reference audio input is compared to main audio input to identify the background noise portion of the main audio signal. A noise reduction algorithm suppresses the identified background noise in the main audio signal. The one or more reference microphones defining the acoustic perimeter may be included in separate microphone devices placed in proximity to the main desktop phone, microphones within other nearby desktop telephone devices, or a combination of both types of devices.

Abstract: An actuator for controlling the operation of an apparatus comprises a panel (16) and an acoustic sensor (26). The panel provides a partition within a building structure, while the acoustic sensor is adapted to detect acoustic waves propagating through the panel. When a user exerts pressure against the panel, the sensor detects the acoustic waves that are formed and emits signals for controlling the operation of an apparatus, such as a doorbell, a light source, a television, a sound system, a ventilation system, a window blind, a radio or an alarm.

Abstract: A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from an output of a microphone that measures ambient audio. The anti-noise signal is combined with source audio to provide an output for a speaker. The anti-noise signal causes cancellation of ambient audio sounds that appear at the microphone. A processing circuit estimates a level of background noise from the microphone output and sets a power conservation mode of the personal audio device in response to detecting that the background noise level is lower than a predetermined threshold.

Abstract: A noise controller includes a first control unit that outputs a control signal for outputting sound for reducing noise to a first speaker, a first characteristic circuit that generates a signal by performing convolution, using a transfer characteristic from the second speaker to a second sound collector, on a control signal output from the first control unit to a second speaker, a subtractor that subtracts the signal generated by the first characteristic circuit from an output signal of a second sound collector and outputs a resultant signal. The first control unit generates the control signal to be output to the first speaker while the output signal from the subtractor serves as a reference signal so that an output signal of the first sound collector is minimized, and outputs the control signal to the first speaker.

Abstract: An active noise reduction device is used with a secondary noise source that generates a secondary noise and an error signal source that outputs an error signal corresponding to a residual sound caused by interference between the secondary noise and a noise. A ?-adjustment unit calculates a step-size parameter for updating a filter coefficient of an adaptive filter by multiplying a standard step-size parameter by a ratio of a standard representative input value corresponding to amplitude of a signal to a representative input value corresponding to the amplitude of the signal.

Abstract: A noise estimation control system may limit increases of a stored background noise estimate in response to a detected noise feedback situation. The system receives an input audio signal detected within a space, and a reference audio signal that is transmitted by a speaker as an aural signal into the space. A signal processor processes the input audio signal and the reference audio signal to determine a coherence value based on an amount of the aural signal that is included in the input audio signal. The signal processor also calculates an amount to adjust the stored background noise estimate based on the coherence value and a determined background noise level of the input audio signal.

Abstract: A sound field control apparatus appropriately presenting desired sound in a listening area and sufficiently reducing the sound in a nearby area without constraint on configuration arrangement includes: a listening compensation filter processing a signal from a sound source using a control characteristic and outputting the signal to a second speaker; and a control filter processing the signal from the listening compensation filter using a control characteristic and outputting the signal to a first speaker. The characteristic of the control filter is a first previously-set characteristic whereby reproduced sound from the second speaker is reduced by reproduced sound from the first speaker at a first control position. The characteristic of the listening compensation filter is a second previously-set characteristic whereby sound having a predetermined target acoustic characteristic is presented by reproduced sounds from the first and second speakers at a second control position.

Abstract: A piezoelectric sound transducer using a piezoelectric plastic material is developed from a substrate layer and a layer of a piezoelectric plastic material deposited thereon, the piezoelectric plastic layer covering the substrate layer not completely but having openings.

Abstract: An active noise control device detects composite vibration of a vibration transmitting route to which both vibration of a rotating body caused by generation or transmission of drive force of a vehicle and vibration of a wheel generated by contact between the wheel and a road surface are transmitted. A first reference signal for defining a reference waveform of a canceling sound for canceling vibration noise in a vehicle interior is generated based on the composite vibration. The component of the canceling sound for canceling vibration noise of the rotating body is removed from the first reference signal to generate a second reference signal for defining a reference waveform of the canceling sound for canceling vibration noise of the wheel. The canceling sound is outputted based on the second reference signal.

Abstract: A noise control device includes the following structural elements. A signal memory records both of a noise signal supplied from a noise microphone and an error signal supplied from an error microphone. A filter coefficient calculator calculates a fixed filter coefficient of a control filter by using data recorded in the signal memory. A filter coefficient renewing section renews, at a given timing, a filter coefficient set at a fixed filter in a control filter to a filter coefficient read out from the filter coefficient calculator.

Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.

Abstract: A hearing protector comprises two protective muffs (1) with passive noise damping, a microphone (3) disposed in at least the one protective muff (1) and connected to an analog and a digital signal processing device (5 and 7, respectively), the signal processing devices (5, 7) being connected to loudspeakers (4) disposed interiorly in the protective muffs (1) for extinguishing interior noise in the protective muffs (1). The analog signal processing device is rehearsed for processing non-repeatable noise, while the digital signal processing device is rehearsed for processing repeatable noise. According to the invention, a further microphone (8) is disposed outside the passive noise damping and is connected to the digital signal processing device for tracing and locking onto repeatable noise.

Abstract: Reflection cancelling boundary microphones are described in which a microphone capsule is mounted within the pressure zone of a vibrating surface and the microphone is tuned in such a way as to cancel pressure waves reflected by the surface while admitting pressure waves generated by the vibration of the surface. One embodiment of the invention includes a microphone capsule configured to be mounted within the pressure zone of a vibrating surface. In addition, the microphone is tuned to cancel pressure waves reflected by the surface while admitting pressure waves generated by the vibration of the surface.

Abstract: An active noise reduction device is used with a secondary noise source that generates a secondary noise and an error signal source that outputs an error signal corresponding to a residual sound caused by interference between the secondary noise and a noise. A ?-adjustment unit calculates a step-size parameter for updating a filter coefficient of an adaptive filter by multiplying a standard step-size parameter by a ratio of a standard representative input value corresponding to amplitude of a signal to a representative input value corresponding to the amplitude of the signal.

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

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: The headset comprises two earpieces each having a transducer for playing back the sound of an audio signal and received in an acoustic cavity defined by a shell having an ear-surrounding cushion. The active noise control comprises, in parallel, a feedforward bandpass filter receiving the signal from an external microphone, a feedback bandpass filter receiving as input an error signal delivered by an internal microphone, and a stabilizer bandpass filter locally increasing the phase of the transfer function of the feedback filter in an instability zone, in particular a waterbed effect zone around 1 kHz. A summing circuit delivers a weighting linear combination of the signal delivered by these filters together with the audio signal to be played back. Control is non-adaptive, with the parameters of the filters being static.

Abstract: Disclosed, in general, are devices that provide an air-tight chamber over a sound source while absorbing all fields of sounds from the sound source. In general, the devices feature: an anechoic chamber that is configured to receive a sound source in an air-tight manner; and an anechoic channel that is in fluid communication with the ambient atmosphere. Suitably, the anechoic chamber is adapted to capture air containing sound energy generated by the sound source, and distribute the air about an internal surface area on the inside of the chamber, wherein the internal surface area is sufficiently large to dampen or otherwise absorb the sounds energy. Preferably, the air is directed from the anechoic chamber through an anechoic channel extending therefrom to the ambient to further dampen or absorb the sound energy. In one configuration, the outer wall of the apparatus is configured to reflect ambient sounds.

Abstract: In an audio system having acoustic echo, incoming audio signals are decorrelated by applying both phase modulation and amplitude modulation. The resulting decorrelated signals are rendered by loudspeakers, and co-located microphones generate outgoing audio signals that include echo signals caused by the existence of acoustic echo paths from the loudspeakers to the microphones. Acoustic echo cancellation (AEC) is applied to the outgoing audio signals to reduce the amount of echo in those signals. Decorrelating the incoming audios signals using amplitude modulation in addition to phase modulation can increase the effectiveness of the AEC processing. In a frequency-domain implementation, the amount of amplitude modulation can be different for different frequency bands (e.g., greater amplitude modulation for lower frequencies corresponding to human speech and/or no amplitude modulation for higher frequencies).

Abstract: A Device Echo Cancellation system and process are disclosed. According to one embodiment, the present invention comprises an input audio source that is converted to a digital signal via a PCM convertor where the amplitude of the input audio source is sampled at regular intervals and translated into digital PCM audio. The resulting PCM audio is analyzed for its dynamic and harmonic content. Corrective sound waves are then generated that reduce the repetition of the original audio (echoes). The corrective sound waves are applied to the repeating audio echoes to create corrected sound waves. The corrected audio is then outputted from the system.

Abstract: Provided are an interested audio source cancellation method, a voice recognition method using the interested audio source cancellation method, computer-readable recording media recording programs for implementing the methods, and a voice recognition apparatus for performing voice recognition by using the interested audio source cancellation method. The interested audio source cancellation method includes steps of: receiving input mixed signals from two microphones and transforming the signals into a time-frequency domain through short-term Fourier transformation; setting an interested audio source cancellation vector for cancelling an interested audio source signal from the input mixed signals in the time-frequency domain; and generating a mixed noise signal by cancelling the interested audio source signal from the input mixed signals by using the interested audio source cancellation vector.

Abstract: Systems and methods are described to reduce undesired audio. An adaptive noise cancellation unit receives a main signal and a reference signal. The main signal has a main signal-to-noise ratio; the reference signal has a reference signal-to-noise ratio. The reference signal-to-noise ratio is less than the main signal-to-noise-ratio. The adaptive noise cancellation unit reduces undesired audio from the main signal. An output signal from the adaptive noise cancellation unit is input to a single channel noise cancellation unit. The single channel noise cancellation unit further reduces undesired audio from the output signal to provide mostly desired audio. A filter control creates a control signal from the main signal and the reference signal to control filtering in the adaptive noise cancellation unit and to control filtering in the single channel noise cancellation unit.

Abstract: An active noise control (ANC) system may be implemented to both sides of a fan, such that both directions of the noise emitting are treated to reduce the overall noise. The impact on airflow is minimal, and the technique is very effective in a broad range of low frequencies. Passive sound-absorbing materials may be included for attenuation of high frequencies. The resulting quiet fan produces a low level of noise compared to any other device based on fan, which produces the same capacity of airflow. The quiet fan may be incorporated in any mechanical system which requires airflow induction such as: computers, air conditioners, machines, and more.

Abstract: A loudspeaker arrangement in a motor vehicle includes at least one loudspeaker, which is able to be driven by an audio processor, the loudspeaker being arranged as at least one electrostatic planar loudspeaker.

Abstract: Systems and methods are described that perform audio source localization in a manner that provides increased robustness and responsiveness in the presence of acoustic echo. The systems and methods calculate a difference between a signal level associated with one or more of the audio signals generated by a microphone array and an estimated level of acoustic echo associated with one or more of the audio signals. This information is then used to determine whether and/or how to perform audio source localization. For example, a controller may use the difference to determine whether or not to freeze an audio source localization module that operates on the audio signals. As another example, the audio source localization module may incorporate the difference (or the estimated level of acoustic echo used to calculate the difference) into the logic that is used to determine the location of a desired audio source.

Abstract: Health-sensing and health-action devices and systems are generally described. The health-sensing device may include one or more of a sensor, a filter, and a transmitter. The sensor may be configured to sense one or more factors relating to an indicator of a health related condition or occurrence such as snoring and may include one or more microphone devices, accelerometers, and/or MEMs devices. The filter may be configured to evaluate a signal from the sensor and determine if the indicator has been detected. The transmitter may be arranged for initiating a transmission based on a signal from the filter. The health-action device may be configured for responding to an indicator of a health related condition or occurrence of a user and may include one or more of a receiver, a processor, and a responder. The health-action device may stimulate the user or may cancel the snoring sound.

Abstract: An signal processing apparatus, system and software product for audio modification/substitution of a background noise generated during an event including, but not be limited to, substituting or partially substituting a noise signal from one or more microphones by a pre-recorded noise, and/or selecting one or more noise signals from a plurality of microphones for further processing in real-time or near real-time broadcasting.

Abstract: Instrument amplification systems incorporating reflection cancelling boundary microphones and multiband compression in accordance with embodiments of the invention are disclosed.

Abstract: This document discusses, among other things, systems and methods for active noise cancellation. One example system includes a digital ANC circuit configured to receive first audio information from a first microphone and to produce an a digital anti-noise signal configured to attenuate noise sensed by the first microphone; an analog ANC circuit configured to receive second audio information from a second microphone and to produce an analog anti-noise signal configured to attenuate noise sensed by the second microphone; and wherein the system is configured to receive an intended audio signal and to provide an output signal for a speaker using the intended audio signal, the analog anti-noise signal, and the digital anti-noise signal.

Abstract: A noise cancellation system is provided, for generating a noise cancellation signal to be added to a wanted signal to mitigate the effects of ambient noise. The system comprises: an input, for receiving an input signal representing ambient noise; a detector, for detecting a magnitude of said input signal; and a voice activity detector, for determining voiceless periods when said input signal does not contain a signal representing a voice. The detector is adapted to detect the magnitude of said input signal during said voiceless periods, and the system is adapted to operate in a first mode when said input signal is above a threshold value, and a second mode when said input signal is below the threshold value. The first mode comprises generating a noise cancellation signal with a first magnitude for at least partially cancelling the ambient noise. The second mode comprises generating a noise cancellation signal with a second magnitude that is less than the first magnitude.

Abstract: A method and apparatus for providing simultaneous enhancement of transmission loss and absorption coefficient using activated cavities is presented. A layer of material is provided, and a backing plate having a plurality of cavities on the top surface of said backing plate, is disposed adjacent a top surface of said layer of material. A screen is disposed along the top surface of said cavities on said backing plate and at least one cavity includes an actuator disposed within the cavity and a control system receiving a signal from the microphone and receiving a signal from the accelerometer and providing a drive signal to the actuator to provide an acoustic output to provide simultaneous insertion loss and absorption which serves to minimize a linear combination of the signal from the microphone and the signal from the accelerometer.

Abstract: The active noise control device includes: a signal obtaining section that obtains an electric signal relating to the predetermined sound; a control section that adjusts an amplitude and a phase of the electric signal obtained by the signal obtaining section; a vibrating section having a diaphragm and a vibrator, the vibrator vibrating in accordance with an output from the control section. Because a sound radiated from the diaphragm toward the first region is substantially in opposite phase to that toward the second region, the control section controls the vibrator so that the diaphragm generates a sound that attenuates the predetermined sound in the first region, and causes the predetermined sound to have a desired frequency characteristic in the second region.

Abstract: In an aspect, the invention features an active noise reduction device including an electronic signal processing circuit. The electronic signal processing circuit includes a first input for accepting a first signal, a second input for accepting a second signal, an output for providing a third signal, a feed-forward path from the first input to the output, and a feed-forward controller for determining the control parameter by calculating a control signal using the first signal and the second signal and then using the control signal to determine the control parameter. The feed-forward path includes a fixed compensation linear filter and a variable compensation filter having an input for receiving a control parameter that applies a selected linear filter from a family of linear filters that vary in both gain and spectral shape and are selectable by the control parameter.

Abstract: Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter.

Abstract: An array speaker system canceling non-uniform radiation patterns and a method for implementing the same are provided. The method for canceling non-uniform radiation patterns includes predicting radiation patterns in an array speaker with respect to input signals, generating cancellation signals with respect to at least one region corresponding to non-uniform radiation patterns of the predicted radiation patterns, synthesizing the input signals and the cancellation signals, and outputting the synthesized signals to the array speaker. Sound signals from which non-uniform radiation patterns having distorted sound are cancelled such that a stable sound field having uniform radiation characteristics may be obtained and provided to a listener.

Abstract: A method for processing the signals from two or more microphones in a listening device, and a listening device for conducting the method which has a casing holding the microphones, a signal processing unit which provides an output signal corresponding to the microphone signals and suited to the user's hearing, and a receiver unit for delivering the output signal to the user whereby the signals from the microphones are analyzed in order to detect when the casing of the listening device is being touched, whereby further the signal processing of the signal processing unit changes whenever touching of the casing is detected.

Abstract: An active noise cancellation system includes an adaptive filter, a signal source, an acoustic actuator, a microphone, a secondary path and an estimation unit. The adaptive filter receives a reference signal representing noise, and provides a compensation signal in response to the received reference signal. The signal source provides a measurement signal. The acoustic actuator radiates the compensation signal and the measurement signal to the listening position. The microphone receives a first signal that is a superposition of the radiated compensation signal, the radiated measurement signal, and the noise signal at the listening position, and provides a microphone signal in response to the received first signal. The secondary path includes a secondary path system that represents a signal transmission path between an output of the adaptive filter and an output of the microphone.

Abstract: An Active Noise Control (ANC) for controlling a noise produced by a noise source may include an acoustic sensor (212) to sense a noise pattern and to produce a noise signal corresponding to the sensed noise pattern, an estimator (202) to produce a predicted noise signal by applying an estimation function to the noise signal, and an acoustic transducer (216) to produce a noise destructive pattern based on the predicted noise signal.

Abstract: An acoustic management system configured to compensate for acoustical dampening is provided. The system includes a microphone configured to detect a first acoustic signal from an acoustic environment and a logic circuit. The logic circuit detects an onset of the acoustical dampening between the acoustic environment and the microphone. The logic circuit generates an acoustic damping compensation filter, which is applied to the first acoustic signal to generate a drive signal.

Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Abstract: Adjustments to an electro-acoustic transducer may be made to match the performance of a second electro-acoustic transducer such that a net inertial force generated by movement of the electro-acoustic transducers' diaphragms are substantially zero. Adjustments may include adjusting a moving mass of one of the elector-acoustic transducers. Adjustments may include applying an equalization to one of the electro-acoustic transducers.

Abstract: A method of increasing the intelligibility of an audio broadcast in an at least partially enclosed space from at least one amplified audio source. An input microphone receives an incident audio wavefront at a first position in the at least partially enclosed space. An active noise control system is employed to generate a cancelling audio wavefront having a magnitude substantially equal to the magnitude of incident audio wavefront and a phase substantially opposite to the phase of the incident audio wavefront. The cancelling audio wavefront is broadcast at a second position in the at least partially enclosed space adjacent to a reflective surface of the at least partially enclosed space so as to attenuate the incident audio wavefront substantially at or near the reflective surface in order to reduce reverberations of the incident audio wavefront. In this manner, reverberations which could reduce the intelligibility of the audio broadcast to an audience is reduced.