Abstract: Provided is a sound reproduction device including: a first loudspeaker having directionality utilizing a parametric effect; a second loudspeaker having directionality broader than that of the first loudspeaker; an orientation adjustment unit configured to change an orientation of the first loudspeaker; an information obtaining device configured to obtain positional information of a listener; and a drive controller electrically connected to the first loudspeaker, the second loudspeaker, the orientation adjustment unit, and the information obtaining device, and configured to control the orientation of the first loudspeaker based on the positional information of the listener.

Abstract: Systems and methods for automatic user specific, condition specific communication system intelligibility testing and optimization are provided. The intelligibility of speech for a particular user is determined using a test of intelligibility administered by an interactive voice response (IVR) application running on a communication server. The intelligibility test can be run for a particular user under different conditions. For each user and/or set of conditions, a set of speech signal adjustment parameters can be determined. A set of speech signal adjustment parameters that will enhance the intelligibility of a speech signal for a user are applied when that user is involved in a communication session. The particular set of speech signal adjustment parameters selected can depend on the communication equipment and/or environment associated with the communication session.

Abstract: Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters.

Abstract: The present invention relates to an ear simulator representing an average acoustic ear drum impedance of ears of a population of humans. Another aspect of the invention relates to an ear simulator assembly comprising an ear simulator representing average acoustic ear drum impedance and a detachable ear canal simulator to provide an ear simulator assembly representing an acoustic impedance of a human ear canal or average human canals of the population.

Abstract: An interface for expanding a signal starting from a first sensing signal and a second sensing signal, wherein a receiving intensity measuring element generates an intensity signal; and a selector is controlled to select each time the first sensing signal, the second sensing signal, or a combined signal deriving from a weighted combination of these signals. The selector uses a plurality of thresholds variable as a function of the intensity signal.

Abstract: A headset in-use detector is disclosed. In an exemplary embodiment, an apparatus includes a detector configured to receive a sound signal and an echo signal and generate a detection signal, and a controller configured to determine whether or not a headset is in-use based on the detection signal.

Abstract: The present invention discloses an impedance detecting device capable of detecting the impedance value of a load, comprising: an AC signal generating circuit for generating an AC signal; an output buffer for generating an output voltage and an output current according to the AC signal and the impedance value of the load; a current mirror for generating a mirror current according to the output current; a detection impedance for generating a detection voltage according to the mirror current; a comparing circuit for generating a comparison result by comparing the output voltage with the detection voltage; and a control circuit for adjusting at least one of a current ratio of the mirror current to the output current and the impedance value of the detection impedance according to the comparison result until the comparison result has satisfied a predetermined requirement.

Abstract: An apparatus comprising: an input configured to receive at least two microphone signals associated with at least one acoustic source; an audio source determiner configured to determine from at least part of the at least two microphone signals at least one audio source based on the at least one acoustic source; an audio source direction determiner configured to determine at least one direction associated with the determined at least one audio source; a calibrator configured to calibrate at least one of the at least two microphone signals based on the at least one direction.

Abstract: Apparatus and method to mix first and second audio channels provided to a headset in response to the quantity of audio devices coupled to its connectors, in response to whether any of those audio devices provides more than one audio channel, and in response to their being a complete lack of electric power provided by any audio device and from a power source of the apparatus, itself.

Abstract: Aspects for monitoring audible tones indicative of operational status of each planar in a multiple planar chassis are described. Included in the aspects is the monitoring of a speaker channel of each planar of a plurality of planars in a common chassis for state changes of beep tones. An operational status of a specific planar emitting the beep tones is identified based on the state changes.

Abstract: An audio recording device comprises an audio input unit, a detect unit, and a control unit. The audio input unit includes a connection unit and an audio receive unit. The connection unit is configured to connect with an external pickup device. The audio receive unit is configured to receive the input of an audio signal from the pickup device. The detect unit is configured to detect the connection between the connection unit and the pickup device. The control unit is configured to display information indicating the input state of the audio signal from the pickup device in response to detection of the connection between the connection unit and the pickup device by the detect unit.

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: Apparatus having corresponding methods and computer-readable media comprise: a speaker interface configured to provide audio to a wearable speaker of a headset and to receive speaker return signals generated by from the wearable speaker of the headset; and a user detection module configured to determine a user-related parameter based on the speaker return signals received by the speaker interface from the wearable speaker of the headset.

Abstract: A personal audio device including earspeakers, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal for each earspeaker from at least one microphone signal that measures the ambient audio, and the anti-noise signals are combined with source audio to provide outputs for the earspeakers. The anti-noise signals cause cancellation of ambient audio sounds at the respective earspeakers. A processing circuit uses the microphone signal(s) to generate the anti-noise signals, which can be generated by adaptive filters. The processing circuit controls adaptation of the adaptive filters such that when an event requiring action on the adaptation of one of the adaptive filters is detected, action is taken on the other one of the adaptive filters. Another feature of the ANC system uses microphone signals provided at both of the earspeakers to perform processing on a voice microphone signal that receives speech of the user.

Abstract: A sound system 100 for producing sound on a loudspeaker 130 is provided wherein the temperature of a voice coil of the loudspeaker may be monitored continuously. The sound system comprises a signal generator 150 for generating an evaluation signal which is combined with an input sound signal 120 to obtain a loudspeaker signal and a monitor 140 for monitoring an electric response of the voice coil to the loudspeaker signal. The loudspeaker signal may be adapted 180 to control the temperature of the voice coil in dependency upon the monitored response. The system can determine the temperature of the voice coil 170 even if the input sound signal comprises only little sound energy.

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: An alarm sounder, which incorporates a piezo-electric output transducer, can be silently monitored using a variable frequency square wave. An initial frequency, close to the upper limit of human hearing, is coupled to the sounder. The transducer draws very little current at this initial frequency. The frequency of the square wave is systematically reduced, and the current draw is continually monitored. A high current indicates a low impedance type of fault. A low current throughout the frequency range indicates a potential high frequency type of fault.

Abstract: The present invention concerns an audio amplifier circuit designed to provide an output signal to an audio transducer, said audio amplifier circuit comprising an audio power amplifier designed to receive an audio signal and designed to generate said output signal, a sensor designed to detect an audible sound having at least one noise component, to generate a detected signal. The audio amplifier circuit also includes a processing block configured to receive said detected signal at its input and to generate an off signal at its output, the latter being located at the input of said audio power amplifier. The processing block processes the detected signal according to said input signal to identify said noise component of said detected signal to generate a reference signal. The processing block generates the off signal when the value of said input signal is lower than the value of said reference signal.

Abstract: A processing chip for a digital microphone and related input circuit and a digital microphone are described herein. In one aspect, the input circuit for a processing chip of a digital microphone includes: a PMOS transistor, a resistor, a current source, and a low-pass filter. The described processing chip possesses high anti high-frequency interference capabilities and the described input circuit possesses high high-frequency power supply rejection ratio.

Abstract: An impedance matched resonant circuit uses inductors coupling an audio source and a speaker array in order to reduce feedback reflection induced by an audio signal traveling from the audio source to the array of speakers. An RC circuit could also be coupled to the positive and negative terminals of the speaker array to reduce the slope differential of the audio signal at certain frequencies. The circuit can be packaged together in a single module with switches to activate and deactivate portions of the circuit to alter the effectiveness of the circuit depending upon need.

Abstract: A method and device for distributing audio data to a number of audio playback devices that are connected to a network. One or more network-related performance criteria of a plurality of the audio playback devices are compared. Based on the comparison, at least one of the audio playback devices is designated as a master device and at least one other of the audio playback devices is designated as a slave device. The master device then controls the distribution of audio data to one or more of the slave devices.

Abstract: An audio controller is provided which includes an audio transducer interface that is capable of receiving audio signals from one or more audio transducers each associated with a channel. A channel selector is configured to selectively connect the audio signals to an output of the audio controller. A tone detector receives a tone from a mobile device and the tone is associated with a selected channel or indicating a channel change. The tone causes the channel selector to output audio associated with a channel. A listening device interface transmits the audio output from the channel selector to a listening device. The audio signal is presented to a mobile device, such as a smartphone or table device. Audio signal received by mobile device may be displayed in graphical form, recorded, or processed and one or more modification may be made to the audio signal such as equalization or an offset.

Abstract: A sound field control apparatus includes at least two main microphones; for each main microphone, a set of at least two sub microphones arranged such that the at least two sub microphones are placed in different axis directions about each of the main microphones; a filtering unit; and a filter coefficient calculating unit configured to calculate a filter coefficient for the filtering unit. A filter coefficient used to control sound pressure levels and air particle velocities of an output audio signal is calculated on the basis of a sound pressure level detected by each main microphone and the difference between the sound pressure level detected by the main microphone and that detected by each of the corresponding sub microphones.

Abstract: An approach for supervision of I2S bus audio paths employing a signal derived from a word clock signal of the I2S bus.

Abstract: Embodiments described herein involve selecting a coordinator device for multichannel playback by a bonded zone. The bonded zone may be configured such that playback devices in the bonded zone may each be configured to render different components, such as a right channel component or a left channel component of media content being rendered in synchrony by the bonded zone. Selecting the coordinator device for the bonded zone may involve analyzing a wireless interface performance metric of a first playback device in the bonded zone, receiving a wireless interface performance metric of a second playback device in the bonded zone, comparing the wireless interface performance metrics of the two playback devices, selecting one of the two playback devices as the coordinator device, and causing the selected playback device to become the coordinator, or “primary” player of the bonded zone.

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 that the sealing section of the seal status.

Abstract: An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and/or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

Abstract: Through a cable coupling two portions of a communications headset, conductors of the cable providing electric power from one of the portions to another are caused to do so with one of a selection of different predetermined voltage levels used to control aspects of one or more functions performed by the communications headset, including enabling or disabling functions, perhaps at least partially by controlling the manner in which the provided electric power is used.

Abstract: A microphone and a method for calibrating a microphone are disclosed. In one embodiment the method for calibrating a microphone comprises operating a MEMS device based on a first AC bias voltage, measuring a pull-in voltage, calculating a second AC bias voltage or a DC bias voltage, and operating the MEMS device based the second AC bias voltage or the DC bias voltage.

Abstract: The present disclosure provides a method of determining microphone polarity in a headset. By being able to detect the polarity of the microphone within the headset, a portable electronic device may be compatible with any headset and is not limited to use with a specific headset. By applying a detection signal over one of lines within a headset cable, a return signal may be sensed on one of the other lines to determine which of the lines is the ground line and which line is the microphone line.

Abstract: Systems and methods are disclosed that may be implemented to locally and remotely record, control, monitor and analyze sounds generated in an information handling system environment, e.g., such as in a server environment. Among other things, remote monitoring and observation of sound in a data center environment may be implemented to assist administrators with data center management and forensics analysis related to physical chassis events (e.g., such as chassis access and/or chassis intrusion), as well as providing live streaming, and local technician recorded sound bite capabilities. A sound policy may also be implemented involving server sound optimizations that works in conjunction with system thermal and power algorithms.

Abstract: An image display device includes: a first audio signal input terminal to which a first audio signal is input from a microphone; and a control section adapted to control power supply from a power supply circuit to the microphone based on type information indicative of whether a type of the microphone to be connected to the first audio signal input terminal is a first type which is requiring power supply or a second type which is not requiring power supply.

Abstract: An information presentation device includes an audio signal input unit configured to input an audio signal, an image signal input unit configured to input an image signal, an image display unit configured to display an image indicated by the image signal, a sound source localization unit configured to estimate direction information for each sound source based on the audio signal, a sound source separation unit configured to separate the audio signal to sound-source-classified audio signals for each sound source, an operation input unit configured to receive an operation input and generates coordinate designation information indicating a part of a region of the image, and a sound source selection unit configured to select a sound-source-classified audio signal of a sound source associated with a coordinate which is included in a region indicated by the coordinate designation information, and which corresponds to the direction information.

Abstract: A wireless headset mode and audio mode selectable audio system includes a major audio device, a wireless audio device, and a wireless headset device. The major audio device includes an audio input selection unit, a digital signal processing unit, a speaker driving unit, a wireless transceving unit, a control unit, and a speaker. The wireless audio device includes a first wireless transceving unit, a first control unit, a first speaker driving unit, and a first speaker. The wireless headset device includes a second wireless transceving unit, a second speaker driving unit, a second control unit, and a second speaker. When the wireless headset is turned on, a signal will be sent to the major audio device to turn off the sound output of the speaker of the major audio device and the first speaker of the wireless audio device, thus entering a mute mode.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for performing audio subsystem enhancement. In one aspect, a method includes: receiving a voice search query by an automatic speech recognition (ASR) engine that processes voice search queries for a search engine, wherein the voice search query includes an audio signal that corresponds to an utterance, and a test flag that indicates that an audio test is being performed; performing speech recognition on the audio signal to select one or more textual, candidate transcriptions that match the utterance; generating, in response to receiving the test flag, one or more audio quality metrics using the audio signal; and generating a response to the voice search query by the ASR engine, wherein the response references one or more of the candidate transcriptions and one or more of the audio quality metrics.

Abstract: An audio playing device may operate in a first mode or a second mode. The audio playing device includes a speaker, an earphone jack receiving an earphone plug of an earphone, a sound output driver, a rotation angle detector, and a processor. The sound output driver only drives the speaker when the audio playing device operates in the first mode and only drives the earphone when the audio playing device operates in the second mode. The rotation angle detector generates a first mode change signal when the earphone plug is rotated in the earphone jack along a first direction over a predetermined angle. The processor changes operation of the audio playing device between the first mode and the second mode when the processor receives the first mode change signal from the rotation angle detector.

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

Abstract: 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: Among other things, techniques and systems are disclosed for detecting musical structures, such as downbeats. In one aspect, a method performed by a data processing device includes receiving an input audio signal. The method includes detecting a meter in the received audio signal. Detecting the meter includes generating an envelope of the received audio signal; generating an autocorrelation phase matrix having a two-dimensional array based on the generated envelope to identify a dominant periodicity in the received audio signal; and filtering both dimensions of the generated autocorrelation phase matrix to enhance peaks in the two-dimensional array. The meter represents a time signature of the input audio signal having multiple beats. Additionally, the method includes identifying a downbeat as a first beat in the detected meter.

Abstract: A system for monitoring a building having one or more microphones coupled to a telephone includes a detector configured to detect a triggering event within the building and transmit an activating signal when the triggering event is detected, and a control module configured to receive the activating signal from the detector. The control module is programmed to activate at least one of the one or more microphones to monitor sound when the activating signal is received.

Abstract: Method and devices for testing a headphone with increased sensation are provided. The headphone can filter and amplify low frequency audio signals, which are then sent to a haptic device in the headphone. The haptic device can cause bass sensations at the top of the skull and at both ear cups. The testing system can evaluate the haptic and acoustic sensations produced by the headphone to evaluate if they have been properly assembled and calibrate the headphones if necessary.

Abstract: A band broadening apparatus includes a processor configured to analyze a fundamental frequency based on an input signal bandlimited to a first band, generate a signal that includes a second band different from the first band based on the input signal, control a frequency response of the second band based on the fundamental frequency, reflect the frequency response of the second band on the signal that includes the second band and generate a frequency-response-adjusted signal that includes the second band, and synthesize the input signal and the frequency-response-adjusted signal.

Abstract: A system (10) for providing media and/or communications data to a plurality of users, 5 the system comprising: a plurality of earphone devices (30); and a server system (20) for providing audio data to each of the plurality of earphone devices (30); wherein each earphone device (30) comprises a processor module (80) for allowing 10 two-way digital communication between the earphone device (30) and the server system (20).

Abstract: Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input signal.

Abstract: Techniques for sensing current delivered to a load by a differential output stage, e.g., in a Class D amplifier. In one aspect, voltages across sense resistors coupled in series with first and second branches of the differential output stage are low-passed filtered and digitized. The sense resistors may be coupled in series with the sources of transistors of the first and second branches, wherein the transistors are selectively switchable on and off by input voltage driving voltages. The input driving voltages may correspond to a ternary voltage waveform such that during a given phase, the two transistors coupled in series with the sense resistors may be turned off. Further aspects provide for the first and second branches having cascoded NMOS and/or PMOS transistors, and the sense resistors being provided between a pair of cascoded transistors.

Abstract: An arrangement and method for assessing the audibility and annoyance of at least one distortion component dn(t) with n=1, . . . , N in the output signal p(t) of a device under test, by generating a virtual auralization output signal pA(t) at the output of an auralization system. The output signal pA(t) contains the distortion component dn(t) at an adjustable magnitude according to a scaling factor Sn provided from a control input, and is supplied to a perceptive model and to a reproduction system used by a listener. The auralization system receives the distortion component dn(t) from a separator which receives a test signal xT(t) from the output of a microphone and a reference signal xR(t) from a reference system.

Abstract: A method, apparatus, and computer-readable storage medium for processing a sound signal is provided. The method includes receiving first reference data associated with a positional relationship between reference locations on a first device, receiving second reference data associated with a positional relationship between reference locations on a second device, receiving a reference transfer characteristic, wherein the reference transfer characteristic is based on the first and second reference data, determining, by a processor, an actual transfer characteristic based on acoustic data resulting from a test signal, and calculating, by the processor, a correction coefficient based on a difference between the reference transfer characteristic and the actual transfer characteristic.

Abstract: A periodogram-based wireless microphone signal detection method, which solves the problem of incapability of distinguishing wireless microphone signal from narrow-band interference, includes: acquiring a time-domain digital signal for detection through antenna module, RF front-end module, ADC module and time-domain signal preprocessing module; and performing detection of microphone signal in frequency domain, that is, calculating an average value of periodogram of M segments of the time-domain digital signal for detection, shifting the average value to obtain a shifted vector, estimating a mean and a covariance matrix of the shifted vector, locating a maximum point of the vector, and taking several points on each side of the point as a center to form an information vector; calculating a decision statistical quantity of the information vector using the decision theory, simulating or calculating a threshold in a predefined method, and deciding the signal is narrow-band interference if the decision statistical q

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

Abstract: An audio tuning system including: an interface configured to provide a test signal to a reference device and to a target device; and a controller configured to acquire a result of audio processing in response to the test signal from the reference device and a result of audio processing in response to the test signal from the target device; compare the results; and adjust an audio processing characteristic value of the target device to match that of the reference device based on a result of the comparison. The audio tuning system may automatically tune the audio characteristic of the target device to match the reference device.