Abstract: An audio control system may receive a plurality of audio input signals. A switch or multiplexer may switch one of the plurality of audio input signals from respective audio sources to a power amplifier. A controller in communication with the multiplexer may control switching of the multiplexer. A display may provide the user with information regarding the switching of the audio signal inputs. A user interface may permit the user to manually switch the audio source. A signal detector in communication with an audio input signal may detect a signal from a priority source. When the signal from the priority source is detected, the controller may direct the multiplexer to route the signal from the priority source to the power amplifier.

Abstract: An earphone set that comprises a paired set of earphones to be worn simultaneously by a user is disclosed. Each earphone in the paired set comprises at least one acoustic transducer for producing sound to be heard by the user. At least one of the earphones in the pair set comprises a transceiver circuit and at least one button. The transceiver circuit is for wirelessly receiving and transmitting signals via a wireless network. The at least one button is accessible to the user at an exterior of the at least one earphone. Upon activation of the at least one button by the user, an indication for a song played by the pair of earphones is caused to be transmitted wirelessly, by the transceiver circuit via the wireless network, to a remote server connected to the Internet. The song played by the pair of earphones may be streamed from a digital audio content server connected to the Internet.

Abstract: Measurements of body sounds in the ear canal may be used for many applications. However, reliability is dependent on the ear canal being properly sealed to allow the body sounds to achieve a detectable level. An apparatus is therefore provided for determining a seal quality indication for a seal of an ear canal. An ear canal microphone (201) provides a microphone signal to an input (203) which is coupled to a circuit (205) for generating a first signal from the microphone signal. The first signal may be the same as the microphone signal. A circuit (209) then determines the seal quality in response to the frequency spectrum for the first signal. A frequency transformer (207) may perform a frequency transformation on the first signal to generate a frequency spectrum for the first signal and. The seal quality indication may specifically be generated based on a detection of a low frequency boost.

Abstract: An audio broadcast collecting method includes a plurality of steps. In one step, an audio broadcast collecting system simultaneously receives a single audio broadcast in a plurality of audio systems. In another step, the audio broadcast collecting system outputs from each of the plurality of audio systems separate audio performances corresponding to the single audio broadcast. In yet another step, the audio broadcast collecting system records an audio sample of the separate audio performances outputted by the plurality of audio systems simultaneously to obtain recorded samples from the plurality of audio systems.

Abstract: A testing method or apparatus utilizes multiple frequencies applied to a device under test for measuring newly discovered frequency modulation effects. An embodiment may include a lower frequency signal with a smaller amplitude higher frequency signal to test a dynamic change in frequency response, gain, and or phase. This dynamic test can reveal frequency modulation effects. Another embodiment may include the use of a multiple frequency signal to dynamically induce a time varying phase or frequency distortion for the device that has differential phase distortion. The device's output is then measured with an FM detector to measure a shift in one of the frequencies used in the test signal or to measure frequency modulation effects of any signals, including distortion products, from the device.

Abstract: A microphone inspection method includes the following steps. Firstly, a sound wave from a speaker is received by an under-test microphone and a reference microphone. Consequently, a first characteristic point distribution chart and a second characteristic point distribution chart are created, respectively. Each of the first characteristic point distribution chart and the second characteristic point distribution chart includes plural characteristic points corresponding to respective normalized frequency values. Then, a characteristic point number difference between a number of the characteristic points of the first characteristic point distribution chart and a number of the characteristic points of the second characteristic point distribution chart within a specified normalized frequency value range is calculated. Consequently, the quality of the under-test microphone is judged according to the characteristic point number difference.

Abstract: A hearing protection system including a detection subsystem configured to determine bone conducted sound vibrations and one or more actuators placed proximate a predetermined location on the skull of a user configured to generate cancellation vibrations out of phase with the bone conducted sound vibrations to mitigate the effect of bone conducted sound vibrations on the middle and/or inner ear of a user.

Abstract: A method and an apparatus control sound signal output. The method of controlling a sound signal output in a terminal includes: storing receiver identifications and corresponding sound signal control parameters associated with each other by the terminal; providing a receiver selection interface capable of selecting one from a list of the receiver identifications and the receiver identifications included in the list by the terminal; receiving one receiver identification through the receiver selection interface; converting a digital sound signal into an analog sound signal by applying a sound signal control parameter corresponding to the received receiver identification; and outputting the analog sound signal.

Abstract: The invention relates to a method for operating a public address system with a control center to which loudspeakers are connected via a two-wire loop system that is routed via isolator modules. A fault can be located on the loop system if the control center measures the impedance of the loop system successively from isolator module to isolator module and the impedance of the entire loop system, inputs the measured values as target values into an impedance table, periodically measures the impedance of the entire loop system during operation and compares said impedance with the corresponding target value from the impedance table, generates a fault message upon determining a deviation, and ascertains the fault location by comparing the measured value with the individual target values in the impedance table and displays said fault location.

Abstract: An audio signal processing apparatus includes a tapping detection unit to which an audio signal picked up by a microphone is input and that detects a tapping input operation on the basis of an energy increase/decrease determination process for making a determination as to whether an increase and a decrease of an energy level of low-frequency components of the audio signal occur within a first time period; and a control unit that performs a certain control process that is set for a tapping input operation in response to a tapping input operation being detected by the tapping detection unit.

Abstract: A signal processing device includes: a sound adjustment amount calculation unit which calculates a sound adjustment amount for adjusting sound characteristics of each channel to a predetermined sound characteristic for each channel, using a sound signal that is obtained by collecting the outputs of each channel; an evaluation value calculation unit which calculates a coefficient allocation evaluation value for allocating a size of a filter coefficient necessary for the sound adjustment of the respective channels for each channel, based on the sound adjustment amount that is calculated by the sound adjustment amount calculation unit; and a filter coefficient calculation unit which calculates the filter coefficient for each channel using the coefficient allocation evaluation value that is calculated by the evaluation value calculation unit.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: Disclosed is a three-way call detection system and method for use with a telephone management system. Specifically, a three-way call detection system and method for detecting continuous noise indicative of an attempt to mask a three-way call attempt. The continuous noise detection system and method may be stand alone or used in conjunction with any existing three-way call detection system and method.

Abstract: A method for testing a speaker (212) and microphone (216) of a communication device (110) is provided. The method includes playing a first audio signal through the speaker (212) and determining whether the first audio signal is audible. If the first audio signal is audible, the communication device (110) receives a second audio signal through its microphone (216) and plays the second audio signal through the speaker (212). If the second audio signal is audible the method determines that the microphone (216) is properly functioning. Such a method allows the communication device (110) to be quickly and easily tested.

Abstract: A telecommunications terminal (100) includes an input audio transducer (111), a receiver (101) for receiving an electrical signal representing a transmitted audio signal, a noise energy estimator (117) for estimating an audio noise energy input received by the input audio transducer and an audio enhancement means (108) for applying to the received electrical signal in a selected audio frequency band a gain which is greater than the gain applied to audio frequencies of the received electrical signal outside the selected band, the gain applied being a function of the audio noise energy input.

Abstract: Audio systems for a vehicle and methods of operating an audio device in a vehicle cabin are provided. A method of operating an audio device directs an acoustic signal to at least one vehicle cabin receiver (VCR) of the audio device; measures sound pressure levels (SPLVCM) for acoustic energy received by at least one vehicle cabin microphone (VCM) of the audio device during a time period ?t; calculates a sound pressure level (SPL) dose (SPL_Dose?t) during the time period ?t using the sound pressure levels SPLVCM of the at least one VCM; combines the SPL dose (SPL_Dose?t) with an SPL dose of a previous time period to form a total SPL dose (SPL_Dosetotal); and performs an action to modify operation of the audio device when the total SPL dose is greater than a predetermined threshold.

Abstract: Disclosed are systems and methods for enhancing playback of an audio signal to approximate the listening experience at a recording site. A test signal is emitted into a recording environment and acoustic parameters for the recording environment are calculated according to the test signal and a response to the test signal. An encoded signal including a recording of an audio event in the recording environment and the acoustic parameters is then transmitted to a listening site or affixed to a medium. At the listening site the acoustic parameters are extracted and the recording is enhanced or corrected according to the acoustic parameters. Acoustic parameters for the listening site may also be calculated and the recording may be further enhanced therewith. During playback of the enhanced recording, the sound may be measured and the enhanced recording may be actively enhanced during playback according to the measurement.

Abstract: The present invention relates to a microphone arrangement (M) which has a charge pump (LP), which produces a DC voltage, a transducer (WA), which converts acoustic signals into electrical signals and which is connected to the charge pump (LP), and a control unit (VCLFS), which controls the charge pump (LP) and which adjusts the DC voltage produced by the charge pump (LP).

Abstract: An audio output controlling apparatus and method, and a mobile terminal system using the same control settings of an audio out port according to whether an audio frame includes an audio signal. The audio output controlling method includes receiving an audio frame, determining whether an audio signal is included in the audio frame, and setting, depending on whether the audio signal is included in the audio frame, a power voltage provided to an audio out port. The mobile terminal includes a wireless transceiver; an audio unit, through an audio out port, an audio frame among the wireless data; and an audio output controlling unit to control a power voltage of the audio out port. The apparatus includes a section determining unit; and a power controlling unit to set a power voltage provided to an audio out port according to whether the audio signal is included in the audio frame.

Abstract: The present invention disclosed an all-digital speaker system device based on multi-bit ?-? Modulation, comprising an A/D converter, an interpolation filter, a ?-? modulator, a dynamic mismatch regulator, a differential buffer and a speaker array. The present invention takes advantage of the ?-? modulation technique to effectively reduce the cost and complexity of the hardware realization of the speaker system device, thus realizing the full digitalization of the whole sound transmission link and the integration of the system device with low power dissipation and volume. Furthermore, the system device according to the present invention is provided with a better control ability on the local sound field, thus providing a better practicable method for the voice secret transmission.

Abstract: A circuit for testing a buzzer includes a microphone, an amplifier circuit, a microprocessor circuit, and a display circuit. The amplifier circuit is configured to amplify an analog signal from the microphone to an amplified signal. The microprocessor circuit determines whether a loudness and a frequency of the amplified signal are within proper ranges. The buzzer is qualified if the loudness and frequency of the amplified signal are within the proper ranges. The display unit displays the loudness, the frequency, and a test result.

Abstract: Devices connected with a computer system are interrogated to detect whether they are configured and functioning. In an audio device implementation, a known audio sample is output to a loudspeaker. If the loudspeaker is properly configured and functional, an attached microphone will capture the sound corresponding to the audio sample generated by the loudspeaker. If sound corresponding to the audio sample is detected, an indication is provided to the user that the microphone and loudspeaker are operational. If no sound corresponding to the audio sample is detected, an indication is provided to the user that the loudspeaker is not enabled and additional configuration is required.

Abstract: Techniques for evaluating the audio quality of an audio test signal are disclosed. These techniques provide a quality analysis that takes into account spatial audio distortions between the audio test signal and a reference audio signal. These techniques involve, for example, determining a plurality of audio spatial cues for an audio test signal, determining a corresponding plurality of audio spatial cues for an audio reference signal, comparing the determined audio spatial cues of the audio test signal to the audio spatial cues of the audio reference signal, and determining the audio quality of the audio test signal.

Abstract: A detection circuit of an electronic device, includes a jack circuit, a first switch, a second switch, and a detection end. The jack circuit includes a type detection pin and a connection detection pin. The first switch includes a first control end. The connection detection pin is connected to the first control end to control the first switch on or off according to whether the audio device is connected to the electronic device. The second switch is connected to ground via the first switch. The second switch includes a second control end. The type detection pin is connected to the second control end to control the second switch on or off according to the type of the audio device. The detection end is connected to the first switch via a first resistor, and is connected to the second switch via a second resistor. The detection end is adapted to generate different signals to indicate a connection status of the electronic device and the audio device.

Abstract: Systems and methods for determining the operating condition of multiple microphones of an electronic device are disclosed. A system can include a plurality of microphones operative to receive signals, a microphone condition detector, and a plurality of microphone condition determination sources. The microphone condition detector can determine a condition for each of the plurality of microphones by using the received signals and accessing at least one microphone condition determination source.

Abstract: A segmentation description scheme for video material that includes references to segments of video contained in different programs. The segmentation description scheme may also interact with a presentation device to present the referenced segments to a user contiguously, without first storing the segments respectively referenced from the different programs.

Abstract: A power saving system includes a sensor, a processor, a timer, and a switch. The sensor senses movements in the diaphragm of the loudspeaker. The processor determines whether the loudspeaker is power-on and in use or power-on but not in use. The processor further determines whether a preset period of time has elapsed since the indication of power-on but not in use was first given. If the preset period of time has elapsed, the processor further controls the switch to disconnect the power source from the loudspeaker.

Abstract: A condenser microphone outputs a signal resistive to distortion caused by an excess input without a variation in signal-to-noise ratio. The microphone includes a first condenser microphone unit and a second condenser microphone unit that generate output signals having phases opposite to each other; and a twin variable resistor connected to an output terminal of each of the first and second condenser microphone units. In the microphone, the output signal of the first unit is combined with the output signal of the second unit in a first variable resistor included in the twin variable resistor and the composite signal is applied to an input of the first unit, and the output signal of the first unit is combined with the output signal of the second unit in a second variable resistor included in the twin variable resistor and the composite signal is applied to an input of the second unit.

Abstract: A test signal emitted by an earpiece is compared against the acoustic signal to determine if a sealing section of the earpiece is sealed properly. The degree of acoustic sealing is used to adjust the attenuation level of a sealing section of the earpiece or alert the user of the seal status of the sealing section.

Abstract: In order to solve the above-described problem, the present disclosure provides a system and a method for signal processing with an external smart device using an earphone port. The earphone port includes a speaker terminal and a microphone terminal and receives a signal from an earphone plug of the peripheral device connected to the smart device. The connection recognition unit recognizes a connection between the earphone plug of the peripheral device and the earphone port. The application operating unit receives an executive command of an existing signal processing application through an input module and executes the signal processing application in response to the executive command. The data processing unit performs a predetermined signal analysis on a data signal input from the peripheral device through the microphone terminal according to an operation of the executed signal processing application and outputs a result of the signal analysis through an output module.

Abstract: According to one embodiment, sound processor includes: communication module; outputting module; recording module; display; input module; controller; and calculating module. The controller (i) displays, on display, message prompting user to move a sound input device to position proximate to speaker, (ii) causes the outputting module to output the test sound and causes the recording module to record first sound, (iii) displays, after the first sound is recorded, on the display, message prompting the user to move the sound input device to listening position, and (iv) causes the outputting module to output the test sound and causes the recording module to record second sound. The calculating module finds a first frequency characteristic of the first sound and a second frequency characteristic of the second sound, and calculates, based on a difference between the first and second frequency characteristics, a value for correcting the second frequency characteristic to a target frequency characteristic.

Abstract: The OPTIMUM BROADCAST AUDIO CAPTURING APPARATUS, METHOD AND SYSTEM (“OBAC”) transforms selection request, video feeds, and, audio feeds inputs via OBAC components into synchronized and optimum video and audio outputs. In one embodiment, a processor-implemented method for capturing optimal audio in sports broadcasting is disclosed, comprising: receiving real time data from a first instrumented camera situated in a venue; processing the real time data to determine a first field position in the venue; activating a first microphone from a plurality of microphones in a first microphone array based on the first field position; and sending a first audio selection from the first microphone.

Abstract: An electrical apparatus includes a frame, a speaker connected to the frame, a digital signal processor in communication with the speaker to receive audio data and control data to control the speaker, the digital signal processor connected to the frame, and a lamp base coupler electrically connected to the speaker and receiver, the lamp base coupler detachably connectable to a power source, when the power source is present. A method of steering the diffused sound field includes, broadcasting at least one calibration audio signal through a plurality of speakers (M) in an audio system, receiving the calibration audio signal in a plurality of microphones spaced apart and positioned about at a listening position, and calculating respective relative speaker placement angles relative to the listening position between each of the plurality of speakers in response to receipt of the calibration audio signal in the plurality of microphones.

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: According to one embodiment, a sound signal compensation apparatus includes an input module, a compensation module, and an output module. The input module receives identification information identifying a first frequency with regard to a resonance of an ear closed by an earphone or headphone. The compensation module performs first compensation emphasizing a second frequency on a sound signal, the second frequency being determined based on the identification information or the first frequency. The output module outputs the compensated sound signal. The compensation module is configured to perform the first compensation emphasizing the second frequency, at which emphasis is greater than or equal to 2 dB and less than or equal to 12 dB.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: Systems, methods, and apparatus are provided for equalization preference learning for digital audio modification. A method for listener calibration of an audio signal includes modifying a reference sound using at least one equalization curve; playing the modified reference sound for a listener; accepting listener feedback regarding the modified reference sound; and generating a weighting function based on listener feedback. A listener audio configuration system includes an output providing a sound for listener review; an interface accepting listener feedback regarding the sound; and a processor programming an audio device based on listener feedback.

Abstract: A method of producing a plurality of broadband response files derived from empirical testing on at least one human subject, for convolving with an audio input signal represented as digital samples to produce a stereo output signal (having a left field and a right field) such that the stereo signal emulates the production of said audio signal from a specified audio source location relative to a listening source location. A reference signal is derived for each of a plurality of test positions. An originating signal is derived from at least one reference signal. Each reference signal is deconvolved with the originating signal to produce a broadband response file for each test position. A data storage facility having a plurality of broadband response files stored therein.

Abstract: A method and apparatus for reducing resonance of a loudspeaker by using a model derived from a software simulated loudspeaker are provided. The method includes generating a model to determine resonance characteristics of the loudspeaker based on physical parameters of the audio reproducing device; determining accuracy of the model of the loudspeaker by comparing calculated frequency characteristics of the model of the loudspeaker and measured frequency characteristics of the loudspeaker that is outputting sound; generating a resonance reduction filter based on the model of the loudspeaker according to the determined accuracy; and reducing the resonance characteristics of the loudspeaker by applying the resonance reduction filter to input audio data.

Abstract: An electronic device includes a housing comprising a speaker configured to be positioned adjacent an ear of a user, and a position-sensitive region on the housing that is configured to sense a position of the ear of the user relative to the housing when the ear is positioned adjacent the speaker and to generate an acoustic leakage position signal responsive to the sensed position of the ear of the user. A control unit is in communication with the speaker and the position-sensitive region and is configured to provide an electrical input signal to the speaker. The control unit is further configured to receive the acoustic leakage position signal and to adapt the electrical input signal responsive to the acoustic leakage position signal.

Abstract: The present invention relates to direct voice and instrument monitoring systems and methods of use thereof.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: Techniques are described to determine that an electrical short has occurred on a communication channel based on monitoring a power level of the communication channel with respect to a reference power level. In an implementation, the aviation communication apparatus includes a first voltage source that provides a first voltage input to a first communication channel of a stereo jack for a headset, a voltage monitor that monitors a voltage level of the first communication channel with respect to a reference voltage level of the first communication channel and determines that an electrical short has occurred on the first communication channel based on the monitoring. Accordingly, an electrical short indicator may indicate responsive to the voltage monitor determining that the electrical short has occurred.

Abstract: A method, hearing aid, and computer product for changing the sequence of program positions of hearing programs are provided. The program positions of the hearing programs have a predefinable sequence. Time periods, within which hearing programs are selected, are measured. A new sequence of the program positions is calculated taking the measured time periods into account. The sequence of the program positions after a predefinable time span is modified in accordance with the calculated sequence. As a result hearing aids adapt to changed environments and place favorite programs in the front section of the program positions in accordance with the hearing aid wearer's preference.

Abstract: A control method of sound producing, sound producing apparatus, and portable apparatus are provided. The control method of sound producing is for a portable device, including the following steps. (a) A first detected parameter indicating a detected excursion of a speaker of the portable device is generated. (b) A second audio signal is generated based on a first audio signal, wherein the second audio signal is generated selectively by compensation according to at least the first detected parameter. (c) A gain value is determined according to at least the first detected parameter and a third audio signal is generated based on the second audio signal according to the gain value.

Abstract: A new hearing device is provided with an inertial measurement unit for determination of yaw of the orientation of a user's head, when the hearing device is worn in its intended operational position on the user's head.

Abstract: Various technologies pertaining to localizing multiple mobile computing devices in an indoor environment are described. Pairs of microphone arrays are selectively positioned in an indoor environment. A localization service assigns a frequency and schedule to a mobile telephone, and the mobile telephone begins outputting vibrations at the assigned frequency and in conformance with the assigned schedule. The microphone arrays sense the vibrations, and angles between the microphone arrays, respectively, and the mobile computing device are computed based upon the sensed vibrations. Such angles are subsequently employed to compute the location of the mobile computing device in the indoor environment.

Abstract: An audio monitoring device for testing an audio device is provided. The audio device is placed in a shielding room which contains a signal emitter. The signal emitter emits electromagnetic waves which electronically interfere with the audio device. The audio monitoring device includes a sound collector, a sound transmitting tube, and a sound outputting element. The sound collector is positioned in the shielding room and collects sounds from the audio device produced whilst being subjected to the electromagnetic interference issued by the signal emitter. The sound transmitting tube has one end connected to the sound collector and the other end exposed from the shielding room, to connect to the sound outputting element. The sound transmitting tube transmits the sounds collected by the sound collector to the sound outputting element for analysis.

Abstract: A control and distribution system provides electrical power distribution, audio signal distribution, and control signal distribution to one or more audio components, such as a powered loudspeaker element or a signal conditioning device such as a rack-mounted amplifier. Embodiments allow for the monitoring and/or control of parameters and/or components at or near the endpoint of the system. These parameters or components include low-level parameters associated with the external audio devices, as opposed to merely higher level parameters of the system. The control and distribution system may include an uninterrupted power source for providing power in an online or offline mode to selected components of the external audio devices. In some embodiments, online backup power is provided to low-power components without providing power to amplifiers within the external audio devices.

Abstract: An in-ear monitor that can be customized for particular applications and individuals includes a housing formed from a body and a cover. A dynamic driver is mounted in a cavity in the housing on an angled mounting flange. The dynamic driver is acoustically coupled to a trumpet-shaped sound collector. The trumpet-shaped sound collector is coupled to a main sound bore that exits an opening in a nozzle portion of the body that is inserted into the ear canal of a user. An ambient sound port collects ambient sound and couples it to the sound bore. An additional bass post increases the bass response of the monitor. Ear impressions are used to customize the body of the monitor to the ear of a user and the location of the bass and ambient sound ports can be altered for different applications.