Abstract: A sound signal processing device supplies an output sound signal to an amplification device configured to supply a first sound signal to a high-frequency speaker and a low-frequency speaker. The sound signal processing device includes a high-pass filter, an amplitude limitation circuit, a low-pass filter, and a synthesis circuit. The high-pass filter removes a low-frequency component from an input sound signal to generate a high-frequency sound signal. The amplitude limitation circuit limits an amplitude of the input sound signal at or below a reference value to generate a second sound signal. The reference value corresponds to a clipping voltage of the amplification device. The low-pass filter removes a high-frequency component from the second sound signal to generate a low-frequency sound signal. The synthesis circuit synthesizes the high-frequency sound signal and the low-frequency sound signal to generate the output sound signal.

Abstract: A filterbank, suitable for modifying audio signals with dynamic gains in each band, is constructed so that the perceived latency is small, while a larger group delay is applied at low frequencies to enable higher frequency resolution in the lower frequency bands. The higher group delay at low frequencies is achieved by inserting an all-pass filter into the reconstructed filter response.

Abstract: An object of the present disclosure is to provide a filter generation device and a filter generation method, capable of generating a filter suitable for out-of-head localization processing. A processing device according to an embodiment includes: a frequency characteristics acquisition unit configured to acquire frequency characteristics based on sound pickup signals; a level calculation unit configured to calculate a reference level in the frequency characteristics; a correction unit configured to correct the frequency characteristics so that the frequency characteristics fall within a predetermined level range including the reference level, and thereby calculate corrected characteristics; and a filter generation unit configured to generate a corrected filter based on the corrected characteristics.

Abstract: The present disclosure provides an adaptive equalization method and system for an audio system. The method includes: detecting an input signal and estimating signal energy of the input signal; determining an audio signal status based on the estimated signal energy; performing dynamic gain calculation based on the estimated signal energy and the audio signal status; and using the calculated gain to adjust equalization parameters, thereby performing dynamic equalization processing on the input signal.

Abstract: Systems and methods for audio signal generation may be provided. A method may include obtaining first audio data collected by a bone conduction sensor; and obtaining second audio data collected by an air conduction sensor, the first audio data and the second audio data representing a speech of a user, with differing frequency component. The method may also include generating, based on the first audio data and the second audio data, third audio data, wherein frequency components of the third audio data higher than a frequency point increase with respect to frequency components of the first audio data higher than the first frequency point.

Abstract: A method and system are presented for monitoring measurement of parameters of patterned structures based on a predetermined fitting model. The method comprises: (a) providing data indicative of measurements in at least one patterned structure; and (b) applying at least one selected verification mode to said data indicative of measurements, said at least one verification mode comprising: I) analyzing the data based on at least one predetermined factor and classifying the corresponding measurement result as acceptable or unacceptable, II) analyzing the data corresponding to the unacceptable measurement results and determining whether one or more of the measurements providing said unacceptable result are to be disregarded, or whether one or more parameters of the predetermined fitting model are to be modified.

Abstract: The present technology relates to an information processing apparatus and method, and a program that enable more efficient edit. The information processing apparatus includes a control unit that selects and groups a plurality of objects existing in a predetermined space, and changes the positions of the plurality of the objects while maintaining the relative positional relationship of the plurality of the grouped objects in the space. The present technology can be applied to information processing apparatuses.

Abstract: An audio generation method and system provided in this disclosure can dynamically select a frequency splicing point of an audio signal based on voice quality of a first audio signal and a second audio signal corresponding to each frequency in a frequency domain, divide the frequency domain into a first frequency interval and a second frequency interval, select audio signals of higher voice quality that correspond to each frequency interval for splicing, and obtain a target audio signal after fusion of the first audio signal and the second audio signal, so that voice quality of the target audio signal in each frequency interval in the frequency domain is the best, thereby improving voice quality of the target audio signal after fusion.

Abstract: A display device, a sound producing control method, a method and device for determining parameters. The display device includes a plurality of exciters arranged on the display panel, the display panel producing sound on a screen through the plurality of exciters; wherein, the plurality of the exciters are formed as at least one sound producing unit, each sound producing unit includes at least two exciters, and each exciter in the same sound producing unit is connected to a same audio signal input end, in one sound producing unit, a signal filter is provided on an audio transmission line from the audio signal input end to at least one exciter, and used for low frequency compensation of a sound produced by the entire sound producing unit.

Abstract: This disclosure describes, in part, techniques to process audio signals to lessen the impact that wind and/or other environmental noise has upon the resulting quality of audio signals output on speaker(s) of wireless earbuds or other headphones. For example, the techniques may determine a level of wind and/or other noise in an environment and may determine how best to process the signals to lessen the impact of the noise, such that one or more users that hear audio based on output of the signals hear higher-quality audio. In some instances, the techniques are able to detect levels of wind in a relatively low-power manner and as part of regular operation of the wireless earbuds or other headphones.

Abstract: A vibration control system includes: a reception section configured to receive eccentric motor drive data that is included in a program for an application and is for causing vibration of an eccentric motor vibration device; a vibration device for which a resonance frequency is higher than that of the eccentric motor vibration device and whose amplitude and frequency are controllable; and a vibration data generation section configured to generate vibration data for causing vibration of the vibration device based on the received eccentric motor drive data. The vibration data generation section generates the vibration data, which indicates a second waveform having an envelope with a change trend that correlates to a change trend of an envelope of a first waveform indicated by vibration of the eccentric motor vibration device vibrated by the eccentric motor drive data, the second waveform having a higher frequency than the first waveform.

Abstract: A sound-producing device includes a housing having a front and a top, a first electro-acoustic transducer facing from the front of the housing, a second electro-acoustic transducer facing from the top of the housing, and a third electro-acoustic transducer facing from the top of the housing. There is at least one processor that is configured to, during audio playback, generate a first array using the first and second electro-acoustic transducers, the first array providing a left height component of the audio playback, and generate a second array using the first and third electro-acoustic transducers, the second array providing a right height component of the audio playback.

Abstract: Implementation of the disclosure provides an audio processing method, a device, a terminal and a computer-readable storage medium. The method can include the following. An audio to-be-matched and a reference audio are obtained. A frequency spectrum distribution of the audio to-be-matched and a frequency spectrum distribution of the reference audio are obtained. A target filter set for matching the audio to-be-matched with the reference audio is determined according to the frequency spectrum distribution of the audio to-be-matched and the frequency spectrum distribution of the reference audio. The target filter set is determined as a matching rule for the audio to-be-matched and the reference audio. An audio playing device is compensated by using the matching rule to adjust an audio playing effect of the audio playing device; and an audio is played through the compensated audio playing device.

Abstract: An audio device and method are disclosed. The audio device includes a microphone, a communication circuit, and a processor. The processor implements the method, including: establishing a communication link with an electronic device via the communication circuit, transmitting information related to the communication link to an external audio device via the communication circuit, detecting a first level of noise included in a first audio data collected via the at least one microphone, identifying a second level of noise of the external audio device, when the first level of noise satisfies a predesignated first condition, and controlling the external audio device to transmit second audio data collected by the external audio device to the electronic device, when the second level of noise of the external audio device meets a predesignated second condition different from the predesignated first condition.

Abstract: A method of incorporating environmental acoustic sources into a virtual environment by measuring real environment acoustic sources and locations and incorporating them into a virtual environment with virtual acoustic sources.

Abstract: In a control method of controlling a frequency response of a filter for processing an audio signal, the frequency response is defined by an amplitude characteristic and a phase characteristic. The control method includes: acquiring a target amplitude characteristic, a target phase characteristic, and a latency value; obtaining a first frequency response in accordance with the target amplitude characteristic and the target phase characteristic; obtaining a second frequency response by modifying the first frequency response so that a latency of the second frequency response satisfies the latency value; and obtaining a third frequency response to be set for the filter by correcting the second frequency response, the second frequency response being corrected so as to reduce a difference between (i) an amplitude characteristic of the second frequency response and (ii) the target amplitude characteristic.

Abstract: A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.

Abstract: A display apparatus and a method for controlling the display apparatus are disclosed. The display apparatus includes: a communication interface comprising communication circuitry, a display, a plurality of speakers, and a processor, and the processor is configured to: control the communication interface to receive an acoustic content corresponding to a sound signal of a plurality of channels corresponding to a specified frequency bandwidth, extend a frequency bandwidth of at least one channel of the plurality of channels based on a specified parametric equalizer (PEQ), perform correction of reducing a gain in middle frequency and high frequency bands of at least one channel of the plurality of channels with the extended frequency bandwidth, and output each sound signal of the plurality of corrected channels to the plurality of corresponding speakers so that a sound image of the sound signal is formed in a center area of the display.

Abstract: A method of audio signal processing comprising Hybrid Expansive Frequency Compression (hEFC) via a digital signal processor, wherein the method includes: classifying an audio signal input, wherein the audio signal input includes frication high-frequency speech energy, into two or more speech sound classes followed by selecting a form of input-dependent frequency remapping function; and performing hEFC including, re-coding of one or more input frequencies of the speech sound via the input-dependent frequency remapping function to generate an audio output signal, wherein the output signal is a representation of the audio signal input having a lower sound frequency.

Abstract: A system and method are disclosed for ignoring a wakeword received at a speech-enabled listening device when it is determined the wakeword is reproduced audio from an audio-playing device. Determination can be by detecting audio distortions, by an ignore flag sent locally between an audio-playing device and speech-enabled device, by and ignore flag sent from a server, by comparison of received audio played audio to a wakeword within an audio-playing device or a speech-enabled device, and other means.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization based on variant selection. An example apparatus to equalize audio includes at least one memory, machine readable instructions, and processor circuitry to at least one of instantiate or execute the machine readable instructions to train a neural network model to apply a first audio equalization profile to first audio associated with a first variant of media, and apply a second audio equalization profile to second audio associated with a second variant of media. The processor circuitry is to at least one of instantiate or execute the machine readable instructions to at least one of dispatch or execute the neural network model.

Abstract: Provided are a method and device for recognizing a speaker by using a resonator. The method of recognizing the speaker includes receiving a plurality of electrical signals corresponding to a speech of the speaker from a plurality of resonators having different resonance bands; obtaining a difference of magnitudes of the plurality of electrical signals; and recognizing the speaker based on the difference of magnitudes of the plurality of electrical signals.

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: A device includes one or more processors configured to receive an input audio signal. The one or more processors are also configured to process the input audio signal based on a combined representation of multiple sound sources to generate an output audio signal. The combined representation is used to selectively retain or remove sounds of the multiple sound sources from the input audio signal. The one or more processors are further configured to provide the output audio signal to a second device.

Abstract: A method including obtaining a first clean signal and a first processed signal dependent upon audio input at a first microphone; obtaining a second clean signal and a second processed signal dependent upon audio input at a second microphone; using the first clean signal and the second clean signal to enable further processing of at least the first processed signal and the second processed signal.

Abstract: A method for automated setting of digital processing parameters intended to be applied to digital signals before broadcast by at least one loudspeaker placed in an environment. The method includes a step of determining a set of frequency responses of the environment, an equalization step applied for each subset of at least one loudspeaker and a second equalization step applied to at least one subset of at least one loudspeaker. A device for the implementation of the method is also provided.

Abstract: The present technology relates to an information processing apparatus and method, and a program that enable more efficient edit. The information processing apparatus includes a control unit that selects and groups a plurality of objects existing in a predetermined space, and changes the positions of the plurality of the objects while maintaining the relative positional relationship of the plurality of the grouped objects in the space. The present technology can be applied to information processing apparatuses.

Abstract: The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length La, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length Ls, generating a frame of the output signal.

Abstract: A system includes a chassis. The system also includes a panel extending in a plane, the panel having a first edge and a second edge opposite the first edge. The system further includes an actuator coupled to the panel at a first location, the actuator being adapted to cause vibration of the panel to generate sound waves. The system further includes a coupling assembly mechanically coupling the chassis to the panel at the second edge of the panel. The panel is fixedly connected to the chassis at the first edge to limit displacement of the panel from the chassis at the first edge during vibration of the panel by the actuator, and the coupling assembly allows displacement of the panel from the chassis at the second edge during the vibration of the panel by the actuator.

Abstract: A beat timing generation device including a generation unit for generating, from data of music that has been input, information of timings on which a beat of the music depends and a plurality of pieces of intensity data indicating a power at the timings; a calculation unit for calculating a period and a phase of the beat of the music using the plurality of pieces of intensity data; and a detection unit for detecting a beat generation timing on a basis of the period and the phase of the beat.

Abstract: An example method of operation may include applying a set of initial power and gain parameters for a speaker, playing a stimulus signal via the speaker, determining a sound level at a microphone location and a sound level at a predefined distance from the speakers, determining a gain at the microphone location based on a difference of the sound level at the microphone location and the sound level at the predefined distance from the speaker, and applying the gain to the speaker output.

Abstract: A method for eliminating unstable noise is provided and applicable to a sound recording device and implemented by a codec. The method includes: activating the sound recording device to start recording; setting a suppression duration and a cutoff frequency switching duration according to unstable noise and a DC offset value of the sound recording device; processing a front-end audio of a recorded sound by a filter having a first cutoff frequency to make the unstable noise in the front-end audio quickly converge, and outputting a filtered audio signal; suppressing the filtered audio signal according to the suppression duration to eliminate the unstable noise; and adjusting the first cutoff frequency of the filter to a second cutoff frequency according to the cutoff frequency switching duration, where the first cutoff frequency is greater than the second cutoff frequency. A device for eliminating unstable noise is also provided.

Abstract: Example methods, apparatus and articles of manufacture (i.e., physical storage media) to perform media source detection based on frequency band selection and processing are disclosed. Example meters disclosed herein are to compare a first audio signal from a monitored media device with a second audio signal from a first one of a plurality of media sources in communication with the monitored media device to determine a sequence of match results, the first audio signal associated with media presented by the media device. Disclosed example meters are also to compute a standard deviation of time delays associated with respective ones of the match results. Disclosed example meters are further to determine whether the first one of the media sources is a source of the media presented by the monitored media device based on the standard deviation.

Abstract: Provided are an electric musical instrument system and a control method thereof. The electric musical instrument system includes a musical sound transmitting means and a musical sound receiving means having a first mode and a second mode that consumes less electric power than the first mode. The musical sound transmitting means has a detecting means for detecting rocking of a musical instrument and a transmitting means that transmits an audio signal emitted from the musical instrument to the musical sound receiving means in the first mode and transmits, when detecting rocking of the musical instruments, detected information to the musical sound receiving means in the second mode. The musical sound receiving means has a controlling means that processes the audio signal received in the first mode and cancels the second mode and changes the mode to the first mode when the detected information is received in the second mode.

Abstract: A method for reducing noise includes: obtaining an air conduction noise reduction parameter and a bone conduction noise reduction parameter, the air conduction noise reduction parameter being obtained by integrating an air conduction parameter of the current frame and an air conduction noise parameter of the current frame, and the bone conduction noise reduction parameter being obtained by integrating a bone conduction parameter of the current frame and a bone conduction noise parameter of the current frame; calculating a priori signal-to-noise ratio of air-bone integration according to the bone conduction parameter of the current frame and the air conduction noise parameter of the current frame; and performing noise reduction operation according to the priori signal-to-noise ratio of air-bone integration, the air conduction noise reduction parameter and the bone conduction noise reduction parameter.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Abstract: A circuit device (100) includes a PWM signal output circuit (110) that outputs a PWM signal (SPWM) to a sound outputter (10), and a processing circuit (120) that controls the PWM signal output circuit (110). The frequency band that the sound outputter (10) can output is denoted as an outputtable band, and the frequency band lower than a lower limit of the outputtable band is denoted as a non-output low frequency band. The PWM signal output circuit (110) outputs a PWM signal (SPWM) based on pseudo sound data in which overtones, of a plurality of overtones of a fundamental tone belonging to the non-output low frequency band, that belong to the outputtable band are used.

Abstract: In some embodiments, a method for performing at least one of enhancement, decoding, or rendering of a multichannel audio signal in response to compression feedback or feedback from a smart amplifier. For example, the compression feedback may be indicative of amount of compression applied to each of multiple frequency bands, of the audio signal or an enhanced audio signal generated in response thereto. The enhancement (e.g., bass enhancement) may include dynamic routing of audio content of the input audio signal between channels of an enhanced audio signal generated in response thereto. The enhancement and compression may be performed on a per speaker class basis. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook or laptop computer, tablet, soundbar, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Abstract: An audio system includes at least one sound generation device having at least an acoustic transducer; and an electronic integrated circuit. A communication bus is connected to the at least one sound generation device and is configured to communicate a digital signal comprised of one or more audio streams and control signals. A power bus is connected to the at least one sound generation device. The acoustic transducer includes at least one of the following groups: a membrane and an acoustic modulator; or a membrane, acoustic resonator, and acoustic coupler. The electronic integrated circuit is constructed and arranged to receive the digital signal and generate an analog electric signal to operate the acoustic transducer to generate an audio signal in accordance with the control signal.

Abstract: An audio signal input and output device includes a port that inputs or outputs an audio signal, an interface that receives a specification of a channel or bus to be assigned to the port, and a sender that sends information over a network for assigning the channel or bus based on the specification received by the interface. The information assigns the inputted audio signal to a predetermined input channel of the management device. A speaker emits a sound based on the inputted audio signal.

Abstract: A hearing device configured for audio communication is disclosed. The hearing device comprises a wireless communication unit and a processing unit, the wireless communication unit being configured to receive and transmit audio signals. The processing unit is configured to receive and/or transmit the audio signal from/to the wireless communication unit, process the audio signal in a first frequency band or in a second frequency band, and detect the quality of the audio communication. If the audio signal is processed in the first frequency band, the processing unit is switching to processing in the second frequency band if the quality of the audio communication is below a first predetermined threshold. If the audio signal is processed in the second frequency band, the processing unit is switching to processing in the first frequency band if the quality of the audio communication is above the first predetermined threshold.

Abstract: A method for controlling an FIR filter includes generating, based on operation information, a first and second control data, generating a second amplitude characteristic, and setting filter coefficients of the FIR filter based on the second amplitude characteristic. The first control data indicates an amount of expansion/compression in a frequency axis direction of a first amplitude characteristic that corresponds to a predetermined transfer function that is expressed as a function of an angular frequency, and the amount of expansion/compression is an integer value or a non-integer value. The second control data indicates an amount of shift in the frequency axis direction of the first amplitude characteristic.

Abstract: The invention relates to a method for processing an acoustic input signal, preferably a speech signal, said method comprising the following steps: a) receiving a digital representation (Sin) of an acoustic input signal, b) calculating at least one statistical parameter (P) of the digital representation (Sin) of the acoustic input signal, c) calculating a compression ratio function (CRf) based—on a prescribed constant compression ratio (CRpr), said prescribed constant compression ratio (CRpr) uniformly mapping acoustic input signals of a selected magnitude to acoustic output signals of a selected magnitude, and—on at least one statistical parameter (P) calculated in step b), and d) applying the non-uniform compression ratio function (CRf) according to step c) on the digital representation (Sin) of the acoustic input signal delivering a digital representation (Sout) of an enhanced acoustic output signal.

Abstract: An information processing device includes a first time-frequency converter configured to perform a time-frequency conversion with respect to an input signal, using a window function having a first width, a second time-frequency converter configured to perform a time-frequency conversion with respect to the input signal, using a second window function having a second width smaller than the first width, and a modification processing unit configured to modify an output of the second time-frequency converter, using a frequency analysis result based on an output of the first time-frequency converter.

Abstract: Disclosed are a device and method for wirelessly communicating. The device according to one example embodiment of the present disclosure may comprise a transceiver and a controller connected to the transceiver, wherein the controller is configured to identify at least one additional sample on the basis of a digital signal by using a neural network model and upscale the digital signal by adding the at least one identified additional sample to a plurality of samples of the digital signal.

Abstract: Disclosed are a sound source output apparatus applying a plurality of DACs and an operating method thereof. The sound source output apparatus according to the exemplary embodiment of the present disclosure includes a sound source acquiring unit which acquires a sound source signal; a tag identification processing unit which identifies tag information about the sound source signal; a DAC selection control unit which selects a specific DAC among a plurality of digital analog converters (DACs) based on a tag identification result of the tag information or a user input signal; and a sound source output unit which outputs a sound source which is converted by the selected specific DAC.

Abstract: Provided a robot for assisting hearing of a user, while minimizing an influence on the surroundings. The robot includes a speaker, a microphone configured to recognize a voice, a processor configured to acquire a position of a user's face when a hearing aid command is acquired on the basis of the voice recognized through the microphone, and a driving unit configured to cause the speaker to be moved toward the position of the user's face, wherein the processor acquires a sound which is a target of hearing aid, generates an assistant sound by amplifying a predetermined frequency band of the sound or converting the predetermined frequency band of the sound into a different frequency band, and outputs the assistant sound through the speaker.

Abstract: A method performed by an in-ear headphone that includes a speaker and an internal microphone. The method receives a microphone signal from the internal microphone that indicates a current sound pressure level (SPL) in an ear canal of a user. The current SPL is a result of a control leak from the in-ear headphone into an ambient environment that reduces a SPL in the ear canal between 2 dB and 25 dB at a frequency within a frequency range than if otherwise not present. The method determines an active noise cancellation (ANC) filter based on the microphone signal and generates an anti-noise signal using the ANC filter. The method drives the speaker using the anti-noise signal to reduce the current SPL in the ear canal of the user as much as 25 dB at a frequency within the frequency range.

Abstract: A method for testing a multiple-actuator storage device includes accessing a first storage device comprising at least a first and a second logical unit and concurrently performing a first group of test operations on the first logical unit and a second group of test operations on the second logical unit. The method includes monitoring for an interruption during the performing the group of test operations on the first or the second logical unit, and during the monitoring, if it is determined that there is an interruption on one of the first or second logical units, performing the respective test operations on the other logical unit is temporarily suspended. The method also includes continuing to monitor for the interruption, and upon determining that the interruption is not currently active, resuming concurrently performing the test operations on the first and second logical units.

Abstract: A headphones system includes reference headphones having native audio performance characteristics, and a personal communication device that generates a user interface presenting visual representations of other headphones. In response to user selection of a visual representation of particular headphones, the personal communication device identifies a corresponding current sound profile establishing a mapping between audio performance characteristics of the particular user selected headphones and the native audio performance characteristics, and communicates the current sound profile to the reference headphones.