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 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: 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: Mechanisms for obtaining an audio signal and information relating to the recording conditions of the audio signal are provided, the method comprising: obtaining an audio signal, wherein the audio signal is a voltage signal representation of sound, the voltage signal having been converted from a pressure signal; obtaining first information indicative of at least one objective feature and/or at least one perceptual feature associated with the conversion of the pressure signal into the audio signal in the form of a voltage signal representation; storing the audio signal, the first information, and second information identifying a relationship between the audio signal and the first information; determining an error adjustment for adjusting the audio signal based on the obtained information; applying the error adjustment to the audio signal to create an error-adjusted audio signal; and obtaining an audio signal and information relating to the recording conditions of the audio signal.

Abstract: To provide a microphone unit capable of acquiring a target sound with high accuracy. A microphone unit in accordance with an exemplary embodiment of the present invention includes a plurality of microphones, a microphone substrate on which the plurality of microphones are mounted, and a vibration observation device disposed at roughly a center of gravity of a shape that is formed by connecting centers of certain adjacent microphones among the plurality of microphones.

Abstract: Systems and methods may be used to modify a controllable stimulus generated by a digital audio device in communication with a human user. An input signal is provided to the digital audio device. In turn, the digital audio device sends a stimulus based on that input signal to the human user, who takes an action, usually in the form of an output signal, to characterize the stimulus that the user receives, based on the user's perception. An algorithm, lookup table, or other procedure then determines a difference between the input signal and the output signal, and a perceptual model is constructed based at least in part on the difference. Thereafter, a new value for the parameter of the digital audio device is suggested based at least in part on the perceptual model. This process continues iteratively until the user's optimal device parameters are determined.

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: According to one embodiment, an electronic device including, a display, an audio output module, a transmission module, a first detection module, a second detection module, a third detection module, and a controller configured to control at least one of the timing of the transmission of the audio signal by the transmission module and the timing of the output of the first reproduction output by the audio output module in accordance with the time difference detected by the third detection module, and to switch whether or not to control the timing in accordance with the positional relationship between the electronic device and the partner device.

Abstract: Electronic devices and accessories for electronic devices such as headsets are provided. The electronic devices may produce audio output. The headsets may include earbuds with speakers that play the audio output for a user while the earbuds are located in the user's ears. Circuitry in an electronic device and a headset may be used in evaluating how well the earbuds are sealed to the user's ears. In response to seal quality measurements, informative messages can be generated for the user, overall earbud volume may be increased, balance adjustments may be made to correct for mismatched balance between left and right earbuds, equalization settings may be adjusted, and noise cancellation circuitry settings can be changed. Electrical impedance measurements and acoustic measurements can be used in evaluating seal quality.

Abstract: A vehicle sound system includes a subwoofer system having a subwoofer, a satellite amplifier for driving the subwoofer, a host amplifier for receiving an audio signal, a noise management processing unit for providing, to the host amplifier, information to be used for achieving a desired audio environment, a sensor for providing information to the noise management processing unit concerning the ambient audio environment, a control line between the satellite amplifier and the host amplifier for transmission of control signals to the satellite amplifier, and a signal line between the satellite amplifier and the host amplifier for transmission of audio signals to the satellite amplifier, the satellite controller being configured to cause a diagnostic signal to be placed on the control line, the diagnostic signal being indicative of an operating condition of the subwoofer system.

Abstract: A method and apparatus may be used to perform personalized audio virtualization. The apparatus may include a speaker, a headphone (over-the-ear, on-ear, or in-ear), a microphone, a computer, a mobile device, a home theater receiver, a television, a Blu-ray (BD) player, a compact disc (CD) player, a digital media player, or the like. The apparatus may be configured to receive an audio signal, scale the audio signal, and perform a convolution and reverberation on the scaled audio signal to produce a convolved audio signal. The apparatus may be configured to filter the convolved audio signal and process the filtered audio signal for output.

Abstract: In this system, a terminal A retrieves terminal's input delay time Tia from input of an audio signal to start of transmission of the audio signal, transmission delay time Tn from the start of transmission of the audio signal to reception of the audio signal by a partner terminal B, partner's reception buffering delay time Tbfb from reception of the audio signal to output of the audio signal by a reception buffer portion BFb of the partner terminal B, and partner's output delay time Tob from input of the audio signal to output of the audio signal by an audio reproduction portion DRob of the partner terminal B. The terminal A then sums up these delay times to determine input/output delay time Ttab from the input of the audio signal by the terminal A to the output of the audio signal by the audio reproduction portion DRob of the partner terminal B.

Abstract: The present invention discloses a test device and test method for the noise reduction headphone. The test device comprises: an enclosed cavity, a noise source, a test panel, a measuring microphone and a measure comparison module connected with the measuring microphone. The sound emitted from the noise source is sealed within the enclosed cavity. The test panel can cooperate with the noise reduction headphone to form a coupling cavity in the test. The test panel has a sound guiding hole in the common part with the enclosed cavity for transmitting the sound of the noise source into the interior of the coupling cavity. The test panel also has a mounting hole, and the measuring microphone is mounted on the mourning hole towards the direction of the coupling cavity. The measuring microphone records noise signals before and after the noise reduction function of the noise reduction headphone is activated.

Abstract: An embodiment of the invention provides a calibration control method performed by a microphone system. The microphone system includes a plurality of microphones configured to generate a plurality of microphone signals. First, the microphone system equalize the microphone signals to generate a plurality of equalized microphone signals. Next, the microphone system calculates a set of similarity indicators based on the equalized microphone signals. Then, the microphone system compares the set of similarity indicators with a set of predetermined thresholds to determine whether to calibrate the microphone signals.

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: A mobile device and a method include setting a vibration intensity grade and activating a vibration sensor to detect vibration intensity of the mobile device. The mobile device and the method further include obtaining the output induced voltage amplitude, confirming a vibration intensity grade, and controlling a music adjuster to adjust a playing speed and a playing volume of a music player to the confirmed vibration intensity grade.

Abstract: According to some aspects an electronic device adapted to detect whether it is coupled to an audio system. The electronic device includes a pulse generator adapted to generate a pulse on a ground return line of the electronic device, and a detector adapted to observe if a response signal is received by the electronic device corresponding to the pulse.

Abstract: Audible signals are created and emitted that provide a human user with improved sound localization cues to quickly and efficiently find the emitting device. Different emitted audible signals are sequentially emitted in response to receiving an activation signal. The different audible signals have been observed to efficiently help a human locate emitting devices that are 1) inside a sealed enclosure, such as a box; 2) in close proximity to the user; and 3) that is a large distance from the user.

Abstract: Methods and systems are provided for adaptively managing a plurality of microphones and speakers in an electronic device. A mode of operation of the electronic device may be determined, and operation of at least one speaker may be managed, based on the determined mode of operation. The managing may comprise adaptively switching or modifying functions of the at least one speaker. For example, the at least one speaker may be configured to act as microphone or as vibration detector. Input obtained using the at least one speaker may be utilized in optimizing audio related functions, such as noise reduction and/or acoustic echo canceling.

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: A system and method is described for monitoring the recording quality of a microphone. Some microphones, mostly in fixed installations, already feature a (piezo) sounder built into the microphone housing which is used to assess whether the microphone is functioning properly. At random or fixed intervals, the sounder produces a known signal; the microphone signal is then measured to determine if the microphone is working The disclosed invention comes down to extension of this principle by not only testing if the signal is reproduced at the correct level, but also using the sounder to measure the transmission quality of the microphone. The sounder produces a signal designed for this purpose. Upon playback of this test signal, the microphone signal is analysed to determine the overall transmission quality of the microphone, and evaluate the influence of factors that degrade quality, such as noise, bandwidth limiting, dynamic range and reverberation.

Abstract: A system and method for using an audiogram and audio filters to provide improved audio characteristics for hearing impaired listeners by equalizing the received sound level of the signal across the frequency spectrum. The system and method further provide for modifying an audio signal based on the personal hearing characteristics of a listener to compensate for hearing loss.

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: 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 amplifier includes a compensation unit, an output unit and a calibration unit. The compensation unit generates a compensation signal based on a digital input signal, a digital reference code, a mode signal and a digital approximation code. The output unit generates an output signal based on the compensated input signal. The calibration unit generates the digital approximation code based on the output signal and the mode signal. The digital approximation code includes a plurality of bits that are generated sequentially.

Abstract: Displacement of a moving diaphragm in an electroacoustic transducer is measured by modulating an electrical signal based on changes in capacitance between the voice coil assembly and the magnetic structure resulting from relative motion between the voice coil and the magnetic structure. The modulated electrical signal is demodulated to produce an output signal having a value proportional to the displacement.

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 or spectrum analysis system 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: Apparatus and method for determining an operating state of an earpiece of a personal acoustic device and/or the entirety of the personal acoustic device through tests to determine the current operating state, wherein the tests differ depending on a current power mode of the personal acoustic device, and wherein at least one lower power test is employed during at least one lower power mode.

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: A method of combining at least two audio signals for generating an enhanced system output signal is described.

Abstract: An information processing apparatus that detects an output from a 3-axis acceleration sensor included in an earphone unit worn by a user while the user is in a still state; monitors the output of the 3-axis acceleration sensor while a nodding gesture is performed by the user; detects a time when an angle of the nodding gesture reaches a maximum; and determines an earphone wearing state based on the output from the 3-axis acceleration sensor in the still state and the output from the 3-axis acceleration sensor at the time of detecting the maximum nodding angle.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

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: Various embodiments are directed to cooperation among communications devices having microphones to employ their microphones in unison to provide voice detection with noise reduction for voice communications. A first communications device comprises a processor circuit; a first microphone; an interface operative to communicatively couple the processor circuit to a network; and a storage communicatively coupled to the processor circuit and arranged to store a sequence of instructions operative on the processor circuit to store a first detected data that represents sounds detected by the first microphone; receive a second detected data via the network that represents sounds detected by a second microphone of a second communications device; subtractively sum the first and second data to create a processed data; and transmit the processed data to a third communications device. Other embodiments are described and claimed herein.

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: A microphone test fixture. The test fixture includes a test chamber, an acoustic source, a reference microphone, and an acoustic resistor. The acoustic source is configured to produce sound waves in the test chamber. The reference microphone is positioned to receive the sound waves in the test chamber. The acoustic resistor forms a contiguous space with the test chamber, and is sized to prevent resonances and echoes of the sound waves for a fixed high frequency limit.

Abstract: Methods and systems are taught for testing microphone packages—including measuring non-acoustic noise for the package. A package positioner holds a microphone package such that an acoustic input port of the microphone package is aligned with a plug. An actuator moves the plug relative to the microphone package between a first position—where the plug does not obstruct the acoustic input port of the microphone package—and a second position—where the plug obstructs the acoustic input port and restricts acoustic pressures from entering the microphone package through the acoustic input port. A controller monitors an output of the microphone package and identifies the output as indicative of isolated non-acoustic noise when the plug is in the second position.

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: An information processing apparatus includes a communication unit, an incoming data determining unit, and an audio processing command generator. The communication unit communicates with an external apparatus that splits outgoing audio data and sequentially sends transmission audio signals, each signal having a header and transmission audio data containing some of the split audio data with an appended first counter. The incoming data determining unit determines the existence of continuity-related errors in a received transmission audio signal, on the basis of the first counter, as well as a second counter contained in the header. The audio processing command generator selectively generates an audio processing command on the basis of the determination results in the incoming data determining unit, wherein the audio processing command stipulates audio data playback processing to be conducted in the event of an error.

Abstract: A headset includes first and second earcups each having a front opening adapted to be adjacent to a respective ear of a user and including an electroacoustic transducer, a headband coupled to each of the earcups, and an active noise reduction circuit coupled to the electroacoustic transducers. The headband is configurable between at least two configurations that each press the earcups against the head of the user with different amounts of force. The active noise reduction circuit is configured to determine which of the at least two configurations the headband may be configured in, to provide a different amount of noise reduction when the headband may be configured in each of the different amounts of force, and to automatically transition between the different amounts of noise reduction in response to a change in the configuration of the headband between the at least two configurations.

Abstract: Apparatus and method for detecting insertion of an external audio outputting device in an electronic device. The apparatus comprises an earphone socket into which the external audio outputting device is inserted. The earphone socket includes first and second terminal sets, each terminal set having at least two electrical contacts. A controller determines an insertion condition if each of the terminal sets is shorted, and determines a non-insertion condition if only one terminal set is shorted, or if neither of the terminal sets is shorted.

Abstract: A method for checking availability of audio interface between a display apparatus and an AV apparatus in an AV system, and an AV apparatus using the same are provided. The checking includes transmitting a command to check an audio interface to a second AV apparatus where an audio from a first AV apparatus is output, and receiving a result of checking the audio interface in response to the command from the second AV apparatus. Therefore, a user can recognize whether or not an audio interface has a problem in an AV system.

Abstract: A method and system for providing information to a user positioned in a room with two independent listening areas where, in one area, an audio signal is output compensating for the sound from the other area, and where a position between the two areas is indicated to a user. When the user moves about, the position and size of one listening area may change, and the user may be made aware of any conflict with the other listening area. The areas may be defined by boundaries where a compensation quality is at a threshold quality.

Abstract: The invention relates to a device, system, and method for a headphone that detects whether the device is in use and either automatically powers up the device when in use, or automatically powers down the device when not in use, or both.

Abstract: A content processing apparatus using sound includes a sound processor configured to generate acoustic sound for a locality check when an event in which locality check is to be executed is generated, a speaker configured to output the acoustic sound, a microphone configured to receive response sound output from an external apparatus which receives the acoustic sound, and a controller configured to determine the external apparatus which outputs the response sound as a peripheral apparatus.

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.

Abstract: A multitrack recorder enables performance of easy identification as to whether a certain track is a monaural type or a stereo type, by means of level meters. The multitrack recorder can set tracks 3 and 4, among a plurality of tracks 1 through 4, to either a monaural type or a stereo type. When track types are stored in memory and when level meters are displayed in a display section, a width of each of track number displays and the number of level displays are changed between the monaural type and the stereo type.

Abstract: A method of setting a parameter for an audio apparatus including a delay processing unit which applies a delaying process and an amplifying process on an input audio signal in accordance with the set parameter, includes: performing a measuring process including: causing an outputting device to output measurement sound; and measuring a first impulse response at a sound receiving point; performing a specifying process including: analyzing the measured first impulse response; calculating a second impulse response at the sound receiving point when the outputting device outputs sound in case where a parameter is set in the audio apparatus and an audio signal indicative of the measurement sound is input to the audio apparatus; and specifying a parameter in accordance with the calculated second impulse response; and performing a setting process of setting the specified parameter in the delay processing unit.

Abstract: The invention relates to a method and an electronic device especially for car audio entertainment including a car audio equipment (2) and an other electronic device (4) like a handheld including at least one data connection (5) and at least one control connection (3) between the car audio equipment and the other electronic device, whereby the car audio equipment is able to send data to the other electronic device which is able to respond to the data send by the car audio equipment by sending a data answer, wherein the car audio equipment is able to compare the data send and the data received and generates a control signal to allow the control of the other electronic device like the handheld by the car audio equipment.

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.