Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization based on variant selection. An example apparatus to generate equalization adjustments for an audio signal includes an input feature set generator to generate an equalization input feature set by generating a data structure corresponding to a user input, the data structure including a number of entries identifying a selected variant of music and including the data structure in the equalization input feature set, and a model executor to, in response to obtaining the equalization input feature set, adjust at least one weight of a neural network model to generate the equalization adjustments for the audio signal.

Abstract: A sound collection system includes a sound collection unit, a variable shelter configured to cover the sound collection unit, an optimum form determination unit configured to determine a form of the variable shelter on a basis of an ambient environmental state, and a shelter optimization control unit configured to change the form of the variable shelter on a basis of a determination result by the optimum form determination unit.

Abstract: Upon an attempt to communicate through an audio input source in a voice session, a determination can be made whether an audio quality received from a user through the audio input source is below an audio quality threshold. In response to determining that the audio quality through the audio input source is below the audio quality threshold, a first voice input can be received from the user from a first position. A second voice input can be received from the user from a second position. A first volume of the first voice input from the first position can be compared to a second volume of the second voice input from the second position. Feedback can be provided to the user indicating whether the user is closer or farther from the audio input source based on the comparison.

Abstract: A system for detecting feedforward instability in a wearable audio device. The audio device includes an electro-acoustic transducer that is configured to develop sound for a user, a housing that holds the transducer, a feedforward microphone that is configured to detect sound outside of the housing and output a microphone signal, and an opening in the housing that emits sound pressure from the transducer that can reach the microphone. A feedforward instability detector is configured to apply two filters to the microphone signal. A first filter passes more energy in a frequency band than does a second filter, to develop a filtered signal. The filtered signal is compared to the microphone signal outside of the frequency band, to develop a comparison signal that is indicative of feedforward instability in the frequency band.

Abstract: A method for determining an actual position of a microphone, comprising: sequentially emitting a first sound signal via a first speaker positioned and a second sound signal via a second speaker positioned; measuring a first elapsed time between the emission of the first sound signal and a detection of the first sound signal by the microphone, and a second elapsed time between the emission of the second sound signal and a detection of the second sound signal by the microphone; determining a first distance between the first speaker and the microphone using the first elapsed time, and a second distance between the second speaker and the microphone using the second elapsed time; determining the actual position of the microphone using the first and second distances and the positions of the first and second speaker; and outputting the actual position.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for wireless audio source switching. One of the methods includes receiving user input selecting a user interface control corresponding to one of a plurality of paired audio source devices with a wireless audio sink device that is a slave device of a first master/slave connection with a first paired audio source device of the plurality of paired audio source devices. The wireless audio sink device initiates a request to become a master device of a second master/slave connection with a second paired audio source device corresponding to the selected user interface control. The wireless audio sink device sends to the second paired audio source device a request to become a slave device of the second master/slave connection. The wireless audio sink device receives audio input from the second paired audio source device and outputs the received audio input.

Abstract: A speaker system may include at least two transducers arranged within an enclosure and horizontally aligned with one another; and a processor configured to apply at least one filter to the transducers to generate beamforming audio content, the processor configured to receive input channels and determine a desired filter impulse response at a first frequency point of the input channels. The processor may also be configured to determine a frequency response of the desired filter impulse response at a first angle, and generate a target function based on the frequency response for application at the first angle.

Abstract: A headphone system including a headphone in wireless communication with a remote-controller to be used in combination for direct communication between a first-user and a second-user. The remote-controller remotely controls functions of the headphone to interrupt a current audio output of the headphone being delivered from an audio-delivering-device. The remote-controller includes an integrated microphone and receive an audio message. Audio from the microphone is effectively instantly delivered to at least one speaker of the headphones via a wireless channel.

Abstract: An acoustic device that reproduces audio signals, the acoustic device including: a first acoustic reproducer that reproduces a signal corresponding to a treble range among the audio signals; a second acoustic reproducer that has a reproduction band from a cutoff frequency on a low frequency side to a cutoff frequency on a high frequency side, and reproduces a signal corresponding to a midrange lower than the treble range among the audio signals; and a harmonic overtone generator that generates a plurality of harmonic overtone signals for a fundamental tone signal corresponding to a specific frequency lower than the cutoff frequency on the low frequency side among the audio signals, wherein at least a part of the plurality of harmonic overtone signals is included in the reproduction band of the second acoustic reproducer.

Abstract: A portable audio input/output device may include one or more openings that extend through a cover of the device and allow acoustic signals outside a housing of the device to reach a microphone disposed within the housing. The opening(s) may be illuminated by a light guide disposed within the housing, which scatters light emitted from lights disposed within the housing. In some instances, a hole may pass through a printed circuit board to allow acoustic signals to be received by the microphone disposed below the printed circuit board. An input/output (I/O) interface module with multiple buttons and inputs may be installed in the hole. The multiple buttons and I/O ports of the I/O interface module may be aligned along an axis vertical relative to the housing and centered with respect to each other.

Abstract: An adaptive identification system, for identifying a propagation system characteristic by an adaptive filter, includes a signal generator that generates an identification input signal including a frequency component of an integer multiple of a fundamental frequency and having a periodicity satisfying a PE condition, a setting unit that sets moving average time to a fundamental period of the identification input signal, and an adaptive algorithm execution unit that uses a moving average value and a diagonal matrix to update a coefficient of the adaptive filter, the moving average value being obtained by calculating a moving average of a cross-correlation vector of a vector of the identification input signal and an observation signal with the moving average time, and the diagonal matrix being obtained by diagonalizing a matrix obtained by calculating a moving average of an autocorrelation matrix of the vector of the identification input signal with the moving average time.

Abstract: A signal processing device includes one or more filter processors that perform sound localization processing of an input acoustic signal and generate output signals; a coefficient setter that sets, for the filter processors, a plurality of filter coefficients for use in the filter processors; an output data selector that selects, out of the output signals subjected to the sound localization processing by the filter processors, output signals to be output to a left speaker and a right speaker; a time measurer that monitors a time for switching the output signals; and a controller that causes the output data selector to select the output signals in accordance with the time for switching the output signals. The plurality of filter coefficients include filter coefficients generated under a plurality of hearing conditions.

Abstract: Endfire linear array microphone systems and methods with consistent directionality and performance at different frequency ranges are provided. The endfire linear array microphone includes a delay and sum beamformer and a differential beamformer. The delay and sum beamformer may produce pickup patterns with good directionality at higher frequency ranges, but cause the pickup patterns to become more omnidirectional at lower frequencies. The differential beamformer may produce pickup patterns with good directionality at lower frequencies. By combining the delay and sum beamformer and differential beamformer within the linear array microphone, the overall directionality of the linear array microphone may be maintained at different frequency ranges while using the same microphone elements.

Abstract: An encoding/decoding apparatus and method for controlling a channel signal is disclosed, wherein the encoding apparatus may include an encoder to encode an object signal, a channel signal, and rendering information for the channel signal, and a bit stream generator to generate, as a bit stream, the encoded object signal, the encoded channel signal, and the encoded rendering information for the channel signal.

Abstract: A microphone array device including microphone capsules and at least one processing unit configured to receive output signals of the microphone capsules, dynamically steer an audio beam based on the received output signal of the microphone capsules, and generate and provide an audio output signal based on the received output signal of the microphone capsules. The processing unit is configured to operate in a dynamic beam mode where at least one focused audio beam is formed that points towards a detected audio source and in a default beam mode where a broader audio beam is formed that covers substantially a default detection area. The microphone array may be incorporated into a conference system.

Abstract: A method for processing an audio signal is provided. In the method, audio signals sent by at least two sound sources are acquired by at least two microphones to obtain multiple frames of original noisy signals of each microphone on a time domain. For each frame, frequency-domain estimation signals of each sound source are acquired according to the original noisy signals of the at least two microphones. For each sound source, the frequency-domain estimation signals are divided into multiple frequency-domain estimation components on a frequency domain. For each sound source, feature decomposition is performed on a related matrix of each frequency-domain estimation component to obtain a target feature vector. A separation matrix of each frequency point is obtained based on target feature vectors and the frequency-domain estimation signals. The audio signals of sounds are obtained based on the separation matrixes and the original noisy signals.

Abstract: A display device is provided to adjust sound balance between an internal speaker of the display device and an external speaker connected with the display device such that a user effectively feels a 3D sound effect. The display device includes a display, an audio output unit configured to output a sound, a wireless communication unit connected with at least one external speaker, and a controller configured to adjust speaker balance between the sound output from the audio output unit and a sound output from the external speaker.

Abstract: A voice enabled device includes a transducer to capture multiple inaudible signals received from multiple ultrasonic speakers and audio recording electronics to process the multiple inaudible signals to generate digital output samples, which are recorded sound data comprising non-linearities from frequency-shifted versions of the multiple inaudible signals to within an audible frequency range. A processing device is to detect, within the recorded sound data, at least a portion of the non-linearities, e.g., via: comparison of the recorded sound data with expected patterns from an audible audio signal generated by human voice; and detection of non-linear variations within the recorded sound data as compared to the expected patterns. In response to the detection, the processing device is further to suppress an action programmed for response to a voice command corresponding to the recorded sound data.

Abstract: The invention relates to a hearing aid a cochlear implant comprising a) at least one input transducer for capturing incoming sound and for generating electric audio signals which represent frequency bands of the incoming sound, b) a sound processor which is configured to analyze and to process the electric audio signals, c) a transmitter that sends the processed electric audio signals, d) a receiver/stimulator, which receives the processed electric audio signals from the transmitter and converts the processed electric audio signals into electric pulses, e) an electrode array embedded in the cochlear comprising a number of electrodes for stimulating the cochlear nerve with said electric pulses, and f) a control unit configured to control the distribution of said electric pulses to the number of said electrodes.

Abstract: The techniques disclosed herein provide apparatuses and related methods for the communication of spatial audio and related metadata. In some implementations, a source provides prerecorded spatial audio that has embedded metadata. A computing device processes the prerecorded spatial audio to generate an audio codec that is segmented to include a first section of audio data and a second section that includes metadata extracted from the prerecorded spatial audio. The generated audio codec may be received by a device that includes an encoder. The encoder may process the generated audio codec to generate audio data that includes the metadata.