Abstract: A system and method may be provided to segment and/or classify an audio signal from transformed audio information. Transformed audio information representing a sound may be obtained. The transformed audio information may specify magnitude of a coefficient related to energy amplitude as a function of frequency for the audio signal and time. Features associated with the audio signal may be obtained from the transformed audio information. Individual ones of the features may be associated with a feature score relative to a predetermined speaker model. An aggregate score may be obtained based on the feature scores according to a weighting scheme. The weighting scheme may be associated with a noise and/or SNR estimation. The aggregate score may be used for segmentation to identify portions of the audio signal containing speech of one or more different speakers. For classification, the aggregate score may be used to determine a likely speaker model to identify a source of the sound in the audio signal.

Abstract: A speaker impedance may be determined by monitoring a voltage and/or current of the speaker. The calculated impedance may be used to determine whether the mobile device containing the speaker is on- or off-ear. The impedance determination may be assisted by applying a test tone low level signal to the speaker. The test tone may be inaudible to the user, but used to determine an impedance of the speaker at the frequency of the test tone. The impedance at that test tone may be used to determine whether a resonance frequency of the speaker is at a frequency corresponding to an on- or off-ear condition. The measured speaker impedance may be provided as feedback to an adaptive noise cancellation (ANC) algorithm to adjust the output at the speaker. For example, when the mobile device is removed from the user's ear, the ANC algorithm may be disabled.

Abstract: A method includes receiving over-the-air (OTA) performance test data measured for a first set of pass computing devices and for a second set of fail computing devices, measuring a first set of training data for the first and second set of computing devices during a plurality of KPI tests, measuring a second set of training data for a particular computing device, determining which ones of the plurality of KPIs qualify as clustering features, and determining a first set of KPI centers, a second set of KPI centers, and a third set of KPI centers, determining a first and a second dissimilarity distance separating the first set and the second set of computing devices from the particular device, respectively. The method further includes determining whether the first dissimilarity distance is greater than second dissimilarity distance to qualify the particular computing device to pass the OTA test.

Abstract: A method for detecting a plurality of available sound output devices when audio data is played back, and for detecting user position information and user direction information is provided. The method includes generating a plurality of pieces of sound information from the audio data, based on at least one of the detected user position information and the user direction information, and distributing each of the plurality of pieces of sound information to a corresponding sound output device from among the plurality of sound output devices.

Abstract: The present invention relates to a microphone test stand comprising a slide assembly with a microphone holder surrounded by a first acoustic sealing member. The slide assembly is movable in a first direction between a first position outside an acoustical chamber with an exposed state of the microphone holder and a second position inside the acoustical chamber in a shielded state of the microphone holder. An electrical connector comprises a set of electrical connector terminals which is connectable to a set of microphone terminals for receipt of a microphone response signal from a microphone assembly arranged in the microphone holder in response to a test sound pressure in the acoustical chamber.

Abstract: A spatial adaptation system for multiple-microphone sound capture systems and methods thereof are described. A spatial adaptation system includes an inference and weight module configured to receive a inputs. The inputs based on two or more input signals captured by at least two microphones. The inference and weight module to determine one or more weight values base on at least one of the inputs. The spatial adaptation system also including a noise magnitude ratio update module coupled with the inference and weight module. The noise magnitude ratio update module to determine an updated noise target based on the one or more weight values from the inference and weight module.

Abstract: An audio signal output from a tone generator 14 is fed into a coil 16 provided in a transducer for vibrating a sound board, so that a sound signal is generated by the vibration of the sound board. A microcomputer 30 obtains temperature values of the coil 16 measured by two different methods, and suspends supply of audio signals to the coil 16 or reduces the amount of audio signals supplied to the coil 16 if the difference between the two coil temperature values is outside a predetermined range. In a case as well where one of the two coil temperature values has risen excessively, the microcomputer 30 suspends supply of audio signals to the coil 16 or reduces the amount of audio signals supplied to the coil 16.

Abstract: Octave errors may be reduced during pitch determination for noisy audio signals. Pitch may be tracked over time by determining amplitudes at harmonics for individual time windows of an input signal. Octave errors may be reduced in individual time windows by fitting amplitudes of corresponding harmonics across successive time windows to identify spurious harmonics caused by octave error. A given harmonic may be identified as either being associated with the same pitch as adjacent harmonics in the given time window or being spurious based on parameters of the fitting function.

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: Systems and methods for improving quality of a call over network (CON) are provided. Call quality may be improved via pathway testing to determine data path quality. This may be utilized to inform buffering lengths, and also may be utilized to choose the data pathway utilized for transmitting the data. Pathway testing may employ collecting microphone data on one device, transmitting it across the various pathways, and then comparing the quality at the endpoint compared to the initial data.

Abstract: The invention concerns an audio signal processing unit including a receiving unit adapted to receive at least a first audio signal and a pilot tone signal, a first audio signal amplifier adapted to obtain at least a second audio signal from an audio signal source by way of an audio signal input, and a pilot tone signal unit adapted to obtain the received pilot tone signal. The pilot tone signal unit is adapted to output a control signal in a first state if the pilot tone signal is received, and to output the control signal in a second state if the pilot tone signal is not received. The first audio signal amplifier is adapted to process the at least second audio signal in dependence on the state of the control signal.

Abstract: A method, an apparatus for adjusting volume and an electronic device using the same is provided. The method comprises: receiving a volume adjusting signal; detecting a current application scene based on a display content of the electronic device after receiving the volume adjusting signal; displaying a volume adjusting interface, the volume adjusting interface including an volume adjusting area which corresponds to the detected current application scene and a selecting area for indicating whether the volume adjusting signal is applied to all application scenes of the electronic device; adjusting volume based on the volume adjusting signal and applying the adjusted volume to the detected current application scene in case that a selected state of the selecting area is in a first state or applying the adjusted volume to all application scenes of the electronic device in case that the selected state of the selecting area is in a second state.

Abstract: A method for determining an audio precompensation controller for an associated sound generating system comprising a total of N?2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L?2 inputs for L input signal(s) and N outputs for N controller output signals, one to each loudspeaker. It is relevant to: estimate (S1), for each one of at least a subset of the N loudspeaker inputs, an impulse response at each measurement position; specify (S2), for each one of the L input signal(s), a selected one of the N loudspeakers as a primary loudspeaker and optionally a selected subset S including at least one of the N loudspeakers as support loudspeaker(s); select (S2) at least one loudspeaker pair, that is required to be symmetrical with respect to the listing position; and specify (S3), for each primary loudspeaker, a target impulse response at each measurement position.

Abstract: Display device ON/OFF detection methods and apparatus are disclosed. Example methods disclosed herein include performing audio code detection on an audio signal corresponding to an output of a presentation device. Such disclosed example methods also include determining a first result based on whether a valid audio code is detected in the audio signal, and determining a second result based on an audio gain level used to adjust the audio signal prior to performing audio code detection on the audio signal. Such disclosed example methods further include combining the first result and the second result to determine a third result indicating whether the presentation device is ON or OFF.

Abstract: Generally, this disclosure provides devices, systems and methods for cancelling an interfering audio signal. The system may include a mobile device including a microphone configured to capture an acoustic audio signal, the acoustic audio signal a combination of the interfering audio signal and a desired audio signal, the desired audio signal generated by a user of the mobile device. The system may also include a wireless communication module incorporated in the mobile device, to receive a reference signal through a side-channel, the reference signal associated with the interfering audio signal. The system may further include an acoustic echo cancellation module incorporated in the mobile device, the acoustic echo cancellation module to cancel the interfering audio signal from the captured acoustic audio signal, the cancellation based on the reference signal.

Abstract: In one embodiment, the invention is a microphone system with an internal test circuit. The system includes a microphone having a housing with an acoustic port, a first transducer, a second transducer, a controller, and a current source. The system also includes an acoustic assembly with a cover and an acoustic pressure source positioned in the cover. When the acoustic assembly is positioned over the acoustic port, an acoustic chamber is formed, and a signal can be applied to the acoustic pressure source such that a first set of measurements can be taken. The acoustic assembly can be removed and replaced with an acoustic cover to take a second set of measurements. Based on the first and second measurements, sensitivities of the first and second transducers can be determined. In another embodiment, the invention provides a method for calibrating the sensitivity of a microphone.

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

Abstract: A Bluetooth transceiver is compatible for usage with a wired headphone, in which the wired headphone has a signal transmission line, at least one speaker, a plug and a control interface for generating a trigger signal to control the operation of a mobile device having Bluetooth wireless communication capabilities. The Bluetooth transceiver includes a headphone jack, a processing unit and a Bluetooth module. The headphone jack is accommodates the plug so that the Bluetooth transceiver is electrically connected to the wired headphone. The processing unit receives and decodes the trigger signal to generate a control code. The processing unit selects a control command according to the control code from a lookup table. The Bluetooth module is electrically connected to the processing unit. The Bluetooth module receives the control command and transmits the control command to the mobile device. The mobile device executes the corresponding functions according to the received control command.

Abstract: The system comprises a master controller and an audio plug connected to the master controller. The audio plug includes at least four contacts electrically connected to corresponding poles of an audio frequency interface. Electrical coupling devices are disposed on circuits which respectively connect a first contact and a second contact with the master controller. The circuits between each electrical coupling devices and the correspondent contact are short circuits.

Abstract: Embodiments of the present teachings disclose method, system, and programs for data driven dynamic logging. Data is received by a logging system where the data is flagged for dynamic logging when one or more dynamic flagging level criteria are satisfied. Data is evaluated by a logging module wrapper, which determines whether to log the data using the default logging level or dynamic logging level, according to whether the data is flagged. A logging module logs information in a logging database according to the default logging level or dynamic logging level. Based on the logged information and detected abnormalities the dynamic flagging level and dynamic logging level are automatically or manually adjusted.

Abstract: An integrated audio signal processing circuit is described for reducing crosstalk noise in an integrated headset switch for a headset having two speakers, a microphone sharing a common ground. The audio signal processing circuit includes two audio amplifiers and also has two terminals configured to be either a microphone connection terminal or a headset ground connection terminal, respectively, depending on the type of the headset. An audio amplifier reference node is coupled to inputs of the audio amplifiers. The audio signal processing circuit has a first switch device responsive to a headset ground selection signal and configured to connect the audio amplifier reference node to the detected headset ground connection terminal. The audio signal processing circuit also has a second switch device responsive to the headset ground selection signal and configured to couple the detected headset ground connection terminal to a ground terminal of the audio processing circuit.

Abstract: Systems and methods for digitally linking multiple microphones and managing microphone signals are provided. Embodiments provide for digitally linking of multiple auto-mixer systems where a large number of microphones are required. In large auto-mixer systems, microphone input channels may be distributed among many devices. Rather than transmitting these signals individually to a central processing device for the auto-mixer, it is most efficient to distribute the audio signal processing functions of the automatic microphone mixer among multiple devices in an audio conferencing system. Subsequently, a smaller number of gated mixes of microphone signals may be transmitted between devices. Thus, embodiments of the present invention act to digitally link multiple microphone signals, arbitrating these signals, in order to enable distributed automatic microphone mixers to behave as a single mixer.

Abstract: A headset test device includes a supporting frame, a drive adjusting system, a head model mechanism and a sensing system. The supporting frame includes a transverse adjusting supporter assembly and a vertical adjusting supporter assembly. The drive adjusting system includes a first motor and a second motor. The transverse adjusting supporter assembly and the vertical adjusting supporter assembly are connected with and are driven by the first motor and the second motor, respectively. The head model mechanism includes a parietal region driven by the second motor to vertically move, and two aural regions driven by the first motor to move close to or away from each other for increasing or reducing a distance between the two aural regions. Each of the two aural regions is equipped with an artificial ear. The sensing system includes a force sensing unit, a pressure sensing unit and a temperature sensing unit.

Abstract: Provided is a method of adjusting a distribution of spatial sound energy, including storing information associated with a sound transfer function from each of speakers of a speaker array to a position of at least one listener, and information associated with the sound transfer function from each of the speakers of the speaker array to a far-field position, and generating at least two sound beams maximizing a far-field sound pressure attenuation with respect to a source signal, based on information associated with the sound transfer function, in order to form a personal sound zone in the position of the at least one listener.

Abstract: In an audio speaker network, setup of speaker location, sound track or channel assignation, and speaker parameters is facilitated by an application detecting speaker locations and prompting a user to input rough room boundaries and a desired listener location in the room. Based on this, optimum speaker locations/frequency assignations/speaker parameters may be determined and output.

Abstract: An automatic audio recovery system in a mobile device is provided. The system includes one or more status generators and an audio path recovery handler. The one or more status generators are configured to monitor operation of audio path elements within the mobile device, and are configured to detect and report one or more abnormal audio operations to an audio status monitor, where, responsive to one or more reports corresponding to the one or more abnormal audio operations, the audio status monitor requests automatic recovery of the audio path elements. The audio path recovery handler is configured to receive an automatic recovery request from the audio status monitor, and is configured to responsively direct reset elements corresponding to the audio path elements to automatically initialize the audio path elements to a stable state.

Abstract: An apparatus may include a signal generation module for generating a signal to test a loudspeaker. The signal generating module may be configured to receive a plurality of inputs identifying parameters relating to a test signal for the loudspeaker, generate a plurality of narrowband signals based on the plurality of inputs, sum the narrowband signals together to produce a broadband test signal, and transmit the broadband test signal to the loudspeaker to generate sound based on the broadband test signal.

Abstract: There is described discovery of a plurality of audio devices, and for the discovered audio devices are determined relative positions thereof and distances therebetween. The determined relative positions and distances are used to select a constellation of audio devices from the discovered plurality. This constellation is selected for playing or recording of a multi-channel audio file so as to present an audio effect such as a spatial audio effect. Channels for the multi-channel audio file are allocated to different audio devices of the selected constellation, which are controlled to synchronously play back or record their respectively allocated channel or channels of the multi-channel audio file. In a specific embodiment the determined distances are used to automatically select the constellation and include distance between each pair of audio devices of the plurality. Several embodiments are presented for determining the distances and relative positions.

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: Techniques for altering audio being output by a voice-controlled device, or another device, to enable more accurate automatic speech recognition (ASR) by the voice-controlled device. For instance, a voice-controlled device may output audio within an environment using a speaker of the device. While outputting the audio, a microphone of the device may capture sound within the environment and may generate an audio signal based on the captured sound. The device may then analyze the audio signal to identify speech of a user within the signal, with the speech indicating that the user is going to provide a subsequent command to the device. Thereafter, the device may alter the output of the audio (e.g., attenuate the audio, pause the audio, switch from stereo to mono, etc.) to facilitate speech recognition of the user's subsequent command.

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: An approach and device for optimizing a sound system by extracting a two-channel stereo source out of a set of multiple channels with unknown frequency responses, crosstalk, time delays and sample rate by employing a digital test sequence that utilizes maximum-length-sequences (MLS) that is fed into the sound system and results in a reconstructed stereo source that may be further used as an input to a signal processor that performs improved speaker and room equalization.

Abstract: A circuit for detecting microphone pin assignment and method thereof is provided. The circuit includes a pin switch unit and a detection unit. The pin switch unit switches the pin assignment of a headset connector. The detection unit receives microphone signals and outputs a test voltage to the microphone. According to the voltage difference between the pins, the pin assignment of the headset is determined and the pin switch unit automatically switches to the correct pins.

Abstract: Voice enhancement and/or speech features extraction may be performed on noisy audio signals. An input signal may convey audio comprising a speech component superimposed on a noise component. The input signal may be segmented into discrete successive time windows including a first time window spanning a duration greater than a sampling interval of the input signal. A transform may be performed on individual time windows of the input signal to obtain corresponding sound models of the input signal in the individual time windows. A first sound model may describe a superposition of harmonics sharing a common pitch and chirp in the first time window of the input signal. Linear fits in time of the sound models over individual time windows of the input signal may be obtained. The linear fits may include a first linear fit in time of the first sound model over the first time window.

Abstract: A packaged MEMS device and a method of calibrating a packaged MEMS device are disclosed. In one embodiment a packaged MEMS device comprises a carrier, a MEMS device disposed on the substrate, a signal processing device disposed on the carrier, a validation circuit disposed on the carrier; and an encapsulation disposed on the carrier, wherein the encapsulation encapsulates the MEMS device, the signal processing device and the memory element.

Abstract: An apparatus and method for interfacing with an earphone are provided. The method includes checking if the earphone is uninstalled promptly after the connection of the earphone is sensed in an external port and if it is sensed that the earphone is uninstalled, continuing to apply a voltage to a MIC_BIAS terminal. The generation of a noise of the MIC_BIAS terminal caused by applying of a periodic voltage resulted from loose insertion of the earphone is prevented.

Abstract: The present invention provides a device and method for diagnosing an audio circuitry, which is applied to a mobile terminal and its earphone. Both of the mobile terminal and its earphone include: a MIC circuit and a sounding circuit, wherein, this device includes: a MIC Circuit Diagnosis Module and a Sounding Circuit Diagnosis Module; A MIC Circuit Diagnosis Module is used for obtaining an amplitude and frequency of a sine wave signal sampled by the MIC circuit, determining whether the MIC circuit is qualified or not according to the amplitude and frequency and outputting the sine wave signal.

Abstract: A method and an electronic device for determining headset line sequence are disclosed in this application, where a third pin and a fourth pin on a headset plug are connected to a direct-current power supply to form a power supply branch, and a bias resistor is connected onto the power supply branch in series. This application obtains a second voltage between the third pin and the fourth pin, compares the second voltage with a preset first threshold, and according to a comparison result, determines a signal type of a signal transmitted by the third pin and a signal type of a signal transmitted by the fourth pin.

Abstract: A system and method for dynamically mixing audio signals may calculate a signal amplitude for each of two or more audio signals. The signal amplitude may be the absolute value of the audio signal. A signal sum may be calculated using each of the two or more signal amplitudes. Each of the two or more signal amplitudes may be smoothed. The signal sum may be smoothed. The smoothing may be a filter or a leaky integrator. A respective mixing gain may be calculated for each of the two or more audio signals using a respective ratio of each of the two or more smoothed signal amplitudes and the smoothed signal sum. Each of the two or more audio signals may be gain adjusted responsive to the respective mixing gain. Each of the two or more gain adjusted audio signals may be mixed to create an output signal.

Abstract: A system includes a housing, a speaker included in the housing, and a first transceiver to wirelessly transmit and receive audio content and control information. The speaker is able to produce audible signals from the audio content received by the transceiver.

Abstract: An audio driver comprising a distortion detection unit coupled between an amplifier and a loudspeaker and configured to detect a distortion signal generated by the loudspeaker while the loudspeaker is being used to deliver an audio signal from the amplifier. A distortion compensation unit configured to compensate for the distortion signal without interfering with the audio signal.

Abstract: Connectivity of a pair of parallel electroacoustic transducers is determined by applying a first test signal on the output line at a frequency where the impedance of the transducers in parallel is less than the impedance of the higher-frequency transducer alone, and observing whether a clip signal is received. If the clip signal is not received, an error indication is output. A second test signal is applied at a frequency where the impedance of the transducers in parallel is less than the impedance of the lower-frequency transducer alone. If the clip signal is not received, the error indication is output. A third test signal is applied at a frequency where the impedance of the transducers in parallel is higher if both transducers are operational than if the higher-frequency transducer is internally short-circuited. If the clip signal is received, a third error indication is output.

Abstract: The invention relates to an optical waveguide system (20) for a display device (1) and to such a display device (1), wherein the optical waveguide system (20) comprises at least one optical waveguide group (20-1, 20-2, 20-3, 20-4, 20-5, 20-6), wherein at least one of the optical waveguide groups (20-1, 20-2, 20-3, 20-4, 20-5) comprises: several optical waveguides (18) made of a material that is transparent to the light of a LED (14), spacers (34) for adjusting a distance of an interspace (30) between a circuit carrier (12) and a front panel (2), and a housing (22) which is made of a material that is intransparent to the light, surrounds each of the several optical waveguides (18) and connects the optical waveguides (18) and the spacers (34) among each other and to each other.

Abstract: Speech portions of a sound model may be identified using various statistics associated with the sound model for voice enhancement of noisy audio signals. A spectral motion transform may be performed on an input signal to obtain a linear fit in time of a sound model of the input signal. Statistics may be extracted from the linear fit of the sound model of the input signal. Speech portions of the linear fit of the sound model of the input signal may be identified by detecting a presence of harmonics as a function of time in the linear fit of the sound model of the input signal based on individual ones of the extracted statistics. An output signal may be provided that conveys audio comprising a reconstructed speech component of the input signal with a noise component of the input signal being suppressed.

Abstract: The method for checking loudspeakers comprises the steps of emitting a pulse over an electrical connection with the loudspeaker; measuring the electrical resistance over the connection; determining the state of the loudspeaker according to the measured resistance; and triggering a corrective action depending on the determined state of the loudspeaker. The claimed method further comprises the steps of associating a plurality of amplification systems to different subsets of the plurality of loudspeakers; assigning different signals to be emitted by different loudspeakers for each of a plurality of control signals; and storing parameters representing the signals to be emitted in each amplification system. When a control signal common to the different amplification systems is received, the amplification systems cause the loudspeakers to emit the different signals and the corrective action comprises memorizing the parameters representing the status of the loudspeaker.

Abstract: A method for determining the temperature of a loudspeaker voice coil comprising the steps of: determining the impedance value of a loudspeaker voice coil at a predetermined evaluation frequency which is greater than 16 kHz and/or in the ultrasonic frequency range; and determining a measure of the temperature of the loudspeaker voice coil based on the impedance value.

Abstract: A loudspeaker drive circuit has a microphone which forms part of an acoustic echo cancellation system. An input signal is processed before application to a loudspeaker driver, and the processing is controlled in dependence on the echo cancellation system performance, such as to control the extent to which the loudspeaker is driven into a non-linear operating region. In this way, the linearity can be controlled so as to provide an excursion limit, without needing a model of the loudspeaker or additional dedicated sensors.

Abstract: Provided is a method and apparatus for reproducing a three-dimensional (3D) sound field. A method of reproducing a 3D sound field may determine a control region based on a wavelength of an excitation frequency, set, based on a loudspeaker array, at least one candidate control point in the control region, and determine a non-uniform control point corresponding to a non-uniform loudspeaker.

Abstract: A system, method and program product for improved techniques for sound management and sound localization is provided. The present invention provides for improving sound localization and detection by inputting a predetermined location's dimensional data and location reference and processing detected sound details, detection device details and the associated location dimensional data as sound localization information for multi-dimensional display. The present invention provides mapping information of sound, people and structural information for use in multiple applications including residential, commercial and emergency situations.

Abstract: An audio setting application of a smart phone acquires tuning data from a tuning data setting server, which meets a search condition specified by a user. The audio setting application performs filtering of tuning parameters included in the tuning data. The tuning parameters are parameters of a plurality of acoustic characteristic items of an audio apparatus, and such parameters are excluded if the parameter cannot be expected to achieve an appropriate effect when the parameter is applied to the audio apparatus.