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: 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: 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 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 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: 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: 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.

Abstract: The embodiments of the present disclosure disclose a method and device for processing an audio signal. A specific embodiment of the method includes acquiring a head image of a target user and an audio signal to be processed; determining head attitude angles of the target user based on the head image, and determining a distance between a target sound source and the head of the target user; and inputting the head attitude angles, the distance and the audio signal to be processed into a preset head related transfer function to obtain a processed left channel audio signal and a processed right channel audio signal, wherein the head related transfer function is used to characterize a correspondence between the head attitude angles, the distance and the audio signal to be processed, and the processed left channel audio signal and the processed right channel audio signal.

Abstract: There is disclosed a device that can record a sound by selecting a directivity direction as desired by the user at the time of recording and can play back the sound by selecting, at the time of playback, a desired directivity direction by the user, separate from the directivity direction specified at the time of recording. The device has microphones, a touch panel for specifying a directivity direction, and a signal processing unit. A recording module performs directivity control in a direction specified on the touch panel and records audio data subjected to effect processing in a memory as directivity-controlled data, and records audio data that are not subjected to directivity control or effect processing in the memory as directivity-control unprocessed data.

Abstract: Disclosed are a method for detecting speech concatenating points and a storage medium. The method includes: acquiring a speech to be detected, and determining high-frequency components and low-frequency components of the speech to be detected; extracting first cepstrum features and second cepstrum features corresponding to the speech to be detected according to the high-frequency components and the low-frequency components; splicing the first and the second cepstrum feature of speech per frame in the speech to be detected in units of frame so as to obtain a parameter sequence; inputting the parameter sequence into a neural network model so as to obtain a feature sequence corresponding to the speech to be detected, wherein the model has been trained, has learned and stored a correspondence between the parameter sequence and the feature sequence; and performing detection of speech concatenating points on the speech to be detected according to the feature sequence.

Abstract: This disclosure provides methods, systems, and apparatuses, for a microphone circuit. In particular, the circuit includes transducer that can sense pressure changes and generate an electrical signal having frequency components in a first frequency range and in a second frequency range higher than the first frequency range. The circuit includes a feedback circuitry that can attenuate frequency components in the first frequency range in the electrical signal and, from it, generate an audio signal. A feedback path circuit includes a low pass filter having a cut-off frequency within the first frequency range, and filters the audio signal to generate a low pass filter signal that includes frequency components in the first frequency range. The low pass filter signal can be used to generate a low frequency pressure signal that corresponds to low frequency pressure changes sensed by the transducer.

Abstract: A computer implemented method for managing a sound emitting device comprising: receiving data associated with operation of the sound emitting device at a predetermined location; processing said data to determine an operating characteristic of that device for that location; comparing the operating characteristic with a predetermined mathematical relationship to determine whether a difference exists; and identifying an input adjustment to correct the difference wherein the input adjustment optionally is within a predetermined range and optionally does not exceed a predetermined maximum increment; wherein the predetermined mathematical relationship is between an input variable and an output variable in respect of the sound emitting device.

Abstract: A method for creating an output soundfield signal from an input soundfield signal, the method including the steps of: (a) forming at least one delayed signals from the input soundfield signal, (b) for each of the delayed signals, creating an acoustically transformed delayed signal, by an acoustic transformation process, and (c) combining together the acoustically transformed delayed signals and the input soundfield signal to produce the output soundfield signal.

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: Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

Abstract: A signal processor for a sound reproduction system which is arranged to perform processing of a sound recording so as to provide input signals for an array of distributed loudspeakers, such that the sound field generated by the loudspeakers results in cross-talk cancellation in respect of multiple listener positions at substantially all frequencies reproduced by the loudspeakers, and wherein the sound field so generated is an approximation of a sound field produced by an Optimal Source Distribution, OSD.

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: An audio processing apparatus includes a linear processing stage and control signal circuitry (CSC). The linear processing stage is configured to receive an input audio signal, and to linearly process the input audio signal based on a user-provided setting of the linear processing stage, so as to produce a linear stage output audio signal. The control signal circuitry (CSC), which is configured to (a) generate a control signal based on (i) the user-provided setting of the linear stage and (ii) a prespecified signal quality preserving criterion, and (b) control the linear stage output audio signal with the control signal, so as to produce a controlled audio signal, in compliance with the signal quality preserving criterion.

Abstract: The present disclosure provides a method and an apparatus for simulating the sound characteristics of the audio processor, a terminal device and a storage medium. The method simulates the sound characteristic of a target audio processor by constructing a nonlinear model. The nonlinear model consists of a pre-filter, a saturation characteristic curve, and a post-filter. A high-level signal is sent to the target sound processor, and the target sound processor correspondingly sends out the first output signal, and then the first output signal is filtered and performed spectrum analysis to acquire the frequency response of the post-filter; And a low-level signal is sent to the target sound processor, and the target sound processor correspondingly sends out the second output signal, and then the second output signal is filtered and performed spectrum analysis to obtain a frequency response product of the pre-filter and the post-filter.

Abstract: At least one exemplary embodiment is directed to a communication device that includes a microphone configured to detect an acoustic signal from an acoustic environment, and a processor, configured to detect an acoustical dampening between the acoustic environment and the microphone, based on a change in a characteristic of the acoustic signal and, responsive to the acoustical dampening, apply a compensation filter to the acoustic signal to form a compensated acoustic signal that is reproduced. Other embodiments are disclosed.

Abstract: A method, system and storage medium for solving echo cancellation failure; method comprises: starting echo cancellation algorithm for convergence of self-adapting filter to stabilize coefficient of filter function (S100); presetting maximum amplitude value for signal intensity that microphone can receive, and detecting if microphone signal lower than maximum amplitude value (S200); prompting user or television system automatically to turn down television volume when microphone received signal larger than or equal to maximum amplitude value, to ensure microphone received signal intensity lower than maximum amplitude value (S300); echo cancellation being carried out without distortion of signals, wherein coefficient of self-adapting filter function W(n) may be continually kept stable, avoiding self-adapting filter converging multiple times due to signal distortion (S400).

Abstract: Noise reduction in a robot system includes the use of a gesture library that pairs noise profiles with gestures performed by the robot. A gesture to be performed by the robot is obtained, and the robot performs the gesture. The robot's performance of the gesture creates noise, and when a user speaks to the robot while the robot performs a gesture, incoming audio includes both user audio and robot noise. A noise profile associated with the gesture is retrieved from the gesture library and is applied to remove the robot noise from the incoming audio.

Abstract: A language proficiency analyzer automatically evaluates a person's language proficiency by analyzing that person's oral communications with another person. The analyzer first enhances the quality of an audio recording of a conversation between the two people using a neural network that automatically detects loss features in the audio and adds those loss features back into the audio. The analyzer then performs a textual and audio analysis on the improved audio. Through textual analysis, the analyzer uses a multi-attention network to determine how focused one person is on the other and how pleased one person is with the other. Through audio analysis, the analyzer uses a neural network to determine how well one person pronounced words during the conversation.

Abstract: Method for carrying out a morphing process, wherein an output parameter relating to the output of an audio signal outputted into an interior via an audio output device is changed.

Abstract: A method for determining a configuration parameter includes: transmitting a detection signal; receiving a reflection signal formed after the detection signal is reflected by a characteristic structure of an ear of a user; adjusting a signal transmission parameter of the detection signal according to a signal difference between the reflection signal and a standard signal until the signal difference meets a first preset condition, to determine the configuration parameter of an earphone.