Abstract: A method of controlling a loudspeaker of an electronic device provides voice coil temperature protection. When a power supply for the electronic device is first activated, a binding step is performed in which the loudspeaker impedance is determined and a temperature (such as ambient temperature) is accurately measured. These binding step measurements are used during subsequent use of the loudspeaker, to make the temperature measurements (based on voice coil impedance) as accurate as possible.

Abstract: The electronic device has audio control for speaker channel based at least in part on an orientation and number of speakers of the electronic device.

Abstract: An audio processing system that includes an audio filter having one or more elements capable of having state, such as a capacitor, an inductor or a delay. A saturation detector is configured to detect saturation of the audio filter and to generate an output when saturation of the filter is detected, such as a switch control signal. A switch is connected to the audio filter and the saturation detector, wherein the state of one or more of the elements of the audio filter is changed when the saturation detector provides the output to the switch, such as when the switch shorts the element and causes the energy stored in the element to be dissipated.

Abstract: A method of modeling the frequency-dependent input-voltage-to-excursion transfer function of a loudspeaker, comprises, for a plurality of measurement frequencies, measuring a voltage and current and deriving an impedance at the measurement frequency. A frequency-dependent impedance function is derived. By additionally using the blocked electrical impedance and a force factor for the loudspeaker, a frequency-dependent input-voltage-to-excursion transfer function can be calculated. The invention provides a modeling approach which is not based on a parametric model, but computes the transfer functions for a set of frequencies separately. As a consequence, it does not require prior knowledge regarding the enclosure (e.g. closed or vented box) and can cope with complex designs of the enclosure.

Abstract: The present publication describes a calibration method and apparatus, in which an electrical calibration signal is formed, an audio signal is formed in the loudspeaker from the calibration signal, the response of the audio signal is measured and analyzed, and the system is adjusted on the basis of the measurement results. The calibration signal is formed in the loudspeaker in such a way that it is essentially a sinusoidal signal, the frequency of which scans at least substantially through the entire audio frequency range.

Abstract: A degree of seal of an ear about a speaker port may be estimated by detecting touch contact between the ear and at least one touch sensor in fixed relation to the speaker port. The degree of seal is estimated based on the detected touch contact. Based upon the estimated degree of seal, the acoustic output of the speaker may be adjusted. The adjustment may compensate for perceived changes to the quality of the acoustic output resulting from the degree of seal. The at least one touch sensor may be a plurality of touch sensors spaced around the speaker port. Each sensor may have a truncated wedge shape, with a narrow end closest to the speaker port. Upon receipt of user input indicative of a high degree of ear seal, a sample of the sensor(s) may be taken and stored for using during future estimation of the degree of seal.

Abstract: An acoustic apparatus which outputs test signals from a plurality of speakers configuring a multichannel reproduction system and adjusts acoustic characteristics by picking up response signals from the speakers using a microphone, includes a test signal storage unit storing test signals output from the speakers; a response signal storage unit storing response signals from the speakers picked up by the microphone; a parameter calculation unit calculating acoustic adjustment parameters including at least a polarity reversal parameter and a phase filter parameter based on the response signals from the speakers stored in the response signal storage unit; and an acoustic adjustment parameter storage unit storing the acoustic adjustment parameters calculated by the parameter calculation unit. The parameter calculation unit calculates the polarity reversal parameter using a low-pass component of the response signals.

Abstract: A method of adjusting a signal processing parameter for a first hearing aid and a second hearing aid forming parts of a binaural hearing aid system to be worn by a user is provided. The binaural hearing aid system comprises a user specific model representing a desired asymmetry between a first ear and a second ear of the user. The method includes detecting a request for processing a parameter change at the first hearing aid, adjusting the signal processing parameter in the first hearing aid, and adjusting a processing parameter for the second hearing aid based on the request for processing parameter change and the user specific model.

Abstract: The invention provides an arrangement and a method for measuring the direct sound wrad radiated by an acoustical source under test (e.g. loudspeakers) under the influence of acoustic ambient noise sources Q1 and reflections at acoustical boundaries (e.g. room walls). An acquisition device measures a state variable pt(rm) of the sound field at a plurality of measurement points rm in a scanning range Gm by a sensor and generates a scanned data set PGm,tQ0Q1. Based on this data set an analyzer determines the coefficients CradQ0 associated with expansion functions which are solutions of the wave equation. An identifier uses the scanned data set pGm,tQ0,Q1 for generating parameter information P for the analyzer which are the basis for separating the direct sound wrad from room reflections Wref and other waves wsec scattered at the surface of the source under test.

Abstract: A system and method are provided for creating a loudspeaker system with low failure rate of sound production. The system uses a loudspeaker with more than one voice coil, a circuit to detect breakage of a voice coil and a switching circuit to steer the system input signal to a remaining good voice coil.

Abstract: A wireless audio system configured to receive audio information wirelessly transmitted by an audio source including first and second wireless transceivers. The first wireless transceiver establishes a bidirectional secondary wireless link with the audio source for receiving and acknowledging receipt of the audio information. The first and second wireless transceivers communicate with each other via a primary wireless link. A wireless audio system including an audio source and first and second wireless transceivers. The first and second wireless transceivers communicate via a primary wireless link. The audio source communicates audio information to the first wireless transceiver via a secondary wireless link which is configured according to a standard wireless protocol. The first wireless transceiver is configured to acknowledge successful reception of audio information via the secondary wireless link.

Abstract: Frequency response correction systems and methods for consumer electronic (CE) devices are disclosed. Applying one or more of disclosed techniques can improve a consumer's listening experience. The frequency response of a CE device can be corrected to remove or attenuate salient, undesirable features of with minimal user interaction. Audio quality can be tuned for optimal or near optimal performance even at maximum or near maximum volume levels substantially without fluctuations, clipping, or any other distortions. In some embodiments, audio response of the CE device can be captured, smoothed, and corrected. Correction parameters can be stored on the CE device.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions.

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

Abstract: A method for determining an inverse filter for altering the frequency response of a loudspeaker so that with the inverse filter applied in the loudspeaker's signal path the inverse-filtered loudspeaker output has a target frequency response, and optionally also applying the inverse filter in the signal path, and a system configured (e.g., a general or special purpose processor programmed and configured) to determine an inverse filter. In some embodiments, the inverse filter corrects the magnitude of the loudspeaker's output. In other embodiments, the inverse filter corrects both the magnitude and phase of the loudspeaker's output. In some embodiments, the inverse filter is determined in the frequency domain by applying eigenfilter theory or minimizing a mean square error expression by solving a linear equation system.

Abstract: The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

Abstract: A laser device, finder of range(s), and an inclinometer are used to figure a sound producing device's position adjustment(s). Laser beam(s) of the laser device and the inclinometer characterize direction(s) of sound produced by the sound producing device. The laser beam(s) can be one or more colors or patterns each characterizing the general direction of sound produced by the sound producing device. A stadium can have many speaker housings each including a speaker, the laser, range finder and the inclinometer. Each speaker housing is mounted on, and remotely adjustable with respect to, an structural member as to the general direction of sound emitted from the speaker housing into the area for stadium seating. The inclinometer's output can be rendered by a monitor seen by an audio engineer who also uses the laser beam's direction to make adjustments to the position of the speaker housing relative to its structural member.

Abstract: A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions.

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: A calibration method for a vibration module includes transmitting a plurality of vibration signals corresponding to a plurality of vibration frequencies to the vibration module and detecting a plurality of input currents or input power levels of the vibration module corresponding to the plurality of vibration frequencies; and determining a vibration point of the vibration module according to the plurality of input currents or input power levels.

Abstract: An audio apparatus is provided. The audio apparatus includes at most one electroacoustic transducer; and an audio controller, coupled to the electroacoustic transducer, for actively controlling the electroacoustic transducer to function as a loudspeaker or a microphone, wherein the loudspeaker converts output electrical signals to output sounds, and the microphone converts input sounds to input electrical signals.

Abstract: A system is provided to protect a loudspeaker (144) by controlling a level of an applied audio signal. A control signal is generated by applying an input audio signal (115) to the collective operations of a gain control system (100). The gain control system (100) uses the input audio signal (115) in conjunction with at least one parameter to derive an estimated stress associated with the loudspeaker (144). The estimated stress is compared with a protection threshold stress (127). If the protection threshold stress is exceeded, a gain applied by a gain component (134) is selectively adjusted to modify the input audio signal (115). The resulting gain-controlled audio signal (116) is employed to drive the loudspeaker (144).

Abstract: A method of configuring an array of speaker elements is disclosed. The method computes a sound that pressure level at various points in the venue and is evaluated by various objective functions. The configuration of a candidate array is changed, for example by the orientation or position of the speakers and the sound field is recalculated. The process is then iterated until an acceptable configuration is found. The real physical array of speakers is then configured in that manner. The method also provides a 3D plot of the sound pressure level displayed against freguency and position in the venue.

Abstract: A device to measure or to estimate the acoustical output of the loudspeaker of a earphone, earplug, headphone or ear muff, and to measure the ambient environmental sounds where appropriate. The loudspeaker is connected to an electronic audio device. The acoustical output of the loudspeaker may be measured by means of a microphone placed in close proximity to the loudspeaker; or estimated by means of measuring the output of the audio amplifier (in the electronic audio device) and using the output/input transfer function that relates the acoustical output of the loudspeaker to the output of the audio amplifier; or estimated by means of the output the digital signal processor (in the electronic audio device) and using the output/input transfer function that relates the acoustical output of the loudspeaker to the output of the digital signal processor.

Abstract: Embodiments for estimating nonlinear distortion and for tuning parameter(s) for boosting sounds are described. A test signal including at least two simultaneous audible tones is generated. One tone is a fundamental tone and others are harmonics of the fundamental tone. The ratio of the number of nonlinear distortion products not coincident with the frequencies of the tones to the number of all the products is, as an example, greater than 0.80. A spectral analysis is performed on the response of a loudspeaker to the test signal. A nonlinear distortion value is estimated by regarding the energy at harmonic frequencies of the fundamental tone signal but not at the frequencies of the tone signals as contribution from the nonlinear distortion. A subjectively correlated measure of nonlinear distortion is obtained for tuning a parameter for boosting low frequency outputs of one or more loudspeakers.

Abstract: Methods and systems for fitting a bone conduction device are provided herein. These methods and systems comprise determining a gain to be used by the bone conduction device in providing signals at a particular frequency. In determining the gain, a fitting system may provide a test sound that is modulated between a first signal provided to a speaker and a second audible signal provided to a bone conduction device. The first and second audible signal may comprise properties such that when the two signals are added together they produce a constant amplitude output. In an embodiment, each of the first and second audible signals may comprise substantially identical frequency characteristics and signal amplitudes, such as, for example, equal amplitude sinusoids centered on the particular frequency for the measurement.

Abstract: Techniques for evaluating at least one relative audio quality parameter of a device, such as a mobile phone, are disclosed. The techniques can include testing in a standard, non-acoustically-isolated environment. The techniques can be used to evaluate whether the device is in compliance with a set of standards.

Abstract: Methods and apparatuses for representing a sound field in a physical space are provided and described as embodied in a system that includes a sound transducer array along with a touch surface-enabled display table. The array may include a group of transducers (multiple speakers and/or microphones). The array may be configured to perform spatial processing of signals for the group of transducers so that sound rendering (in configurations where the array includes multiple speakers), or sound pick-up (in configurations where the array includes multiple microphones), may have spatial patterns (or sound projection patterns) that are focused in certain directions while reducing disturbances from other directions.

Abstract: A sound field controller includes: a sound field generation section which generates an effect sound signal for giving a sound field effect sound to an audio signal; an acquisition section which acquires a measurement signal indicating sound pressure levels of a direct sound and a reflected sound which are collected when a test sound is emitted in a reproduction environment; an identification section which identifies a maximum reflected sound whose sound pressure level is the maximum in a given time period after a collecting timing of the direct sound from the measurement signal; an adjustment section which adjusts the effect sound signal based on a ratio of the sound pressure level of the direct sound to the sound pressure level of the maximum reflected sound; and an output section which outputs the audio signal input to the input section and the effect sound signal adjusted by the adjustment section.

Abstract: In one general aspect, a method can include calculating, at a calibration temperature of a speaker, a calibration parameter through a coil of the speaker in response to a first test signal, and can include sending a second test signal through the coil of the speaker. The method can also include measuring a parameter through the coil of the speaker based on the second test signal, and calculating a temperature change of the coil of the speaker based on the parameter and based on the calibration parameter at the calibration temperature.

Abstract: A testing system tests an audio connection between an audio source and a loudspeaker. The system includes a loudspeaker that converts a reference signal into a sound. An adaptive filter processes the reference signal to minimize an error signal. A decision circuit analyzes the error signal and the received signal to determine signal correlation. When the signals are not correlated, a defect is detected.

Abstract: A method for controlling a maximum signal level output to headphones of a wireless device is provided. The method includes: determining an impedance of the headphones; determining a carrier specific maximum signal level for headphones having the impedance; and, adjusting an audio amplifier of the wireless device coupled to the headphones to restrict the maximum signal level output to the headphones to the carrier specific maximum signal level.

Abstract: Implementations disclosed herein relate to transmitting audio in a target space relative to an electronic device (100, 200) using a directional speaker (103, 106) and non-directional speaker (104, 207). In one implementation, a directional speaker (103, 106) or non-directional speaker (104, 207) is selected to transmit audio based on a target space for the audio. The audio may then be transmitted from the selected speaker.

Abstract: In some embodiments, a method for monitoring speakers within an audio playback system (e.g., movie theater) environment. In typical embodiments, the monitoring method assumes that initial characteristics of the speakers (e.g., a room response for each of the speakers) have been determined at an initial time, and relies on one or more microphones positioned in the environment to perform a status check on each of the speakers to identify whether a change to at least one characteristic of any of the speakers has occurred since the initial time. In other embodiments, the method processes data indicative of output of a microphone to monitor audience reaction to an audiovisual program. Other aspects include a system configured (e.g., programmed) to perform any embodiment of the inventive method, and a computer readable medium (e.g., a disc) which stores code for implementing any embodiment of the inventive method.

Abstract: Disclosed is subject matter that proposes a system and method for a media device to automatically detect the characteristics of an attached speaker. Speakers have many different characteristics, for example, power, impedance, frequency response, etc. With knowledge of the speaker characteristics, audio output can be equalized appropriately, and an amplifier of the media device, for example, can prevent exceeding the maximum power handling capability of the speaker. Described is a device and method for retrieving information about the speaker from a memory that is coupled to the speaker. A media device can read the data from the memory over existing speaker wires. Software and/or hardware in the media device can optimize the output to the attached speaker. Accordingly, the media device can interrogate the speaker directly over speaker wire to obtain the characteristics of the speaker.

Abstract: An audio system that comprises an audio server and individual speaker clients may be connected. Individual speaker clients may or may not be the same physical configuration. When the audio server detects each speaker client, the audio server configures each speaker to a particular purpose. This allows remote configurable speakers to be used for different features. Other audio devices may also be connected to the audio server.

Abstract: A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions.

Abstract: An apparatus comprising a housing, a speaker, and an amplifier. The speaker may be mounted within the housing. A concave portion may be implemented in the housing and may be configured to hold a transformer. An amplifier may be implemented within the housing and may be configured to receive power through a connection to the transformer.

Abstract: A method discloses measuring electroacoustic parameters of transducer. With known voice-coil displacement, voice-coil current, transducer impedance and its stimulus signal as inputs, the five calculation procedures of direct problem, adjoint problem, sensitivity problem, conjugate gradient method, and constraint equations are involved in inversely solving electroacoustic parameters. The presented method has the characteristics of high efficiently, low iterations for computational algorithm, and high accuracy for electroacoustic parameters estimation. Through the numerical result and discussion, the relative errors between estimated and accurate electroacoustic parameters are sufficiently small even with the inclusion of the inevitable measurement errors. These results indicate that the presented method has high feasibility for estimating electroacoustic parameters of a transducer.

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: A MEMS speaker device including a membrane that forms a first capacitor and a second capacitor, respectively, with a top plate and with a bottom plate. The device includes a driving circuit that operates, during a first operating period, to move the membrane into a first position, in which the membrane is close to the bottom plate, and during a second operating period, to move the membrane into a second position, in which the membrane is close to the top plate. The device includes a testing circuit having a measuring circuit, which generates a first signal, based on a capacitance of one of the first capacitor and the second capacitor and a second signal based on a capacitance of one of the first capacitor and the second capacitor; and a comparator, which compares the first and second signals with at least one first electrical reference quantity.

Abstract: An apparatus, method and a computer program are provided. The apparatus is configured to process an electrical output signal from a loudspeaker to detect a user input signal.

Abstract: Audio transducers (headphones, speakers, microphones) inherently do not accurately reproduce the signal presented to them at the input. This can be compensated for by taking into account the transducer characteristics and transforming the input signal using a digital signal processor (DSP) to counteract the inaccuracies. However, for the compensation to take place, the DSP needs to know the characteristics of the transducer. For systems with built-in transducers (like laptops with internal speakers) the characteristics of the internal speakers can be stored on the hard-drive of the laptop and the DSP can read this data and make the appropriate compensations. Because a transducer (headphone, speaker, microphone) has its own characteristics that need to be compensated for separately, a profile is supplied to the DSP either by a database lookup based on an identification made by the user or transducer itself or by profile information stored on the transducer.

Abstract: In a sound output system including: an information processing apparatus; a first output device; and a second output device, the information processing apparatus generates, based on predetermined information processing, a first sound signal to be outputted to the first output device, and a second sound signal that is a sound signal to be outputted to the second output device and has a content different from that of the first sound signal. When headphones are connected to the second output device, the second sound signal is generated such that at least a part of a first sound which is not outputted as a second sound when no headphones are connected, is contained in the second sound, and the second sound is outputted from the headphones.

Abstract: A mobile communication terminal including: a window glass provided on a front side of a terminal main body; a speaker hole formed on an edge of the terminal main body corresponding to an edge of an outer portion of the window glass, exposed to the outside, and formed to correspond to a speaker provided in the terminal main body; and a sensor connected to the speaker, and sensing a direction in which the terminal main body is slanted, wherein the sensor controls the speaker to open or close the speaker hole according to the direction in which the terminal main body is slanted.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: A method and portable terminal for automatically controlling an output volume of audio output of a head phone when the head phone is connected to a portable terminal are provided. The method of controlling an audio output in a portable terminal, includes outputting a test audio signal to a head phone for a preset playback time when the head phone is connected to the portable terminal; collecting neighboring noise interval information and impedance interval information of the head phone while the test audio signal is output to the head phone; determining impedance of the head phone based on the neighboring noise interval information and the impedance interval information; determining a gain of an audio signal based on the determined impedance; and amplifying the audio signal in proportion to the determined gain, and outputting the amplified audio signal to the head phone.

Abstract: The method comprises the determination of an observation vector that comprises only electrical measurements of the voltage (Umes) at the loudspeaker terminals and of the current (i) passing through the loudspeaker, and a state vector (X) whose components comprise: values of linear parameters of the loudspeaker response such as the electrical (Re) and mechanical (Req) resistance, and polynomial coefficients of nonlinear parameters such as the force factor (Bl), the equivalent stiffness (Keq) and the electrical inductance (Le). The voltage and current measurements are applied to an estimator with a predictive filter of the extended Kalman filter incorporating a representation of a dynamic model of the loudspeaker. This filter operates a prediction of the state vector (X) and readjusts this prediction by calculation of an estimate (Uest) of the voltage based on the state vector and on the measured current and comparison of this estimate with the measurement (Umes) of the voltage.

Abstract: A headset is able to be coupled via a cable to an intercom system, is able to be wirelessly coupled to a wireless device via a wireless transceiver of the headset, and is able to be connected to a wired device via another cable. A controller of the headset separately monitors the microphone conductors and audio conductors by which the headset may be coupled to the intercom system to detect whether or not one or both of a communications microphone and an acoustic driver of the headset are coupled to the intercom system, and monitors the operating state of the wireless transceiver to detect whether or not the wireless transceiver is inactive, on standby or in use; and selectively couples a system ground conductor to one of the microphone conductors, and selectively provides a local microphone bias voltage in response to what is observed through such monitoring.