Abstract: An analysis system has at least one monitoring assembly that includes at least one microphone to monitor an acoustic field proximate the monitoring assembly. The at least one microphone produces at least one microphone signal responsive to the acoustic field. The analysis system further includes a data storage device configured to buffer the at least one microphone signal and an audio analysis system configured to analyze a content of the data storage device where the audio analysis system is configured to analyze the content of the buffer to process at least a sound into a response or action.

Abstract: An audio signal processing apparatus is provided. The audio signal processing apparatus includes an input terminal receiving an input audio signal, a processor obtaining a difference between a playback loudness of the input audio signal and a desired loudness thereof and correcting a frequency band spectrum of an output audio signal for each of a plurality of frequency bands based on the difference between the playback loudness and the desired loudness of the input audio signal and a relationship between a loudness and a sound pressure for each of the plurality of frequency bands, and an output terminal outputting the output audio signal. The playback loudness is a loudness of the output audio signal when the input audio signal is output without the correction.

Abstract: A multi-channel and multi-zone audio environment is provided. Various inventions are disclosed that allow playback devices on one or more networks to provide an effective multi-channel and a multi-zone audio environment using timing information. According to one example, timing information is used to coordinate playback devices connected over a low-latency network to provide audio along with a video display. In another example, timing information is used to coordinate playback devices connected over a mesh network to provide audio in one or more zones or zone groups.

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: An equalizer and a method of controlling same are provided. The equalizer includes a memory storing an EQ value set for a plurality of music attributes and storing a general-purpose EQ value; and a processor configured to: obtain an input music signal; calculate a plurality of probability values for the plurality of music attributes by analyzing attributes of the input music signal based on a convolutional neural network; calculate a moderate index between the plurality of probability values; generate an EQ value based on the plurality of probability values and the moderate index; and perform equalizing by applying the generated EQ value to the input music signal.

Abstract: A driving sound library provides driving sounds classified in various themes to a user by analyzing frequency characteristics and temporal characteristics of sound sources classified into categories and determining a chord corresponding to each sound source based on the frequency characteristics and the time characteristics. Modulated sound sources in which each sound source is modulated are generated by applying a chord corresponding to each sound source. The driving sound sources are generated using the sound sources and the modulated sound sources as input data. Pitches of the driving sound sources are then changed based on the driving sound sources and engine sound orders corresponding to an engine RPM of a preset range. Scores for each theme for the driving sounds are generated.

Abstract: A noise reduction method and apparatus for an on-board environment, an electronic device and a storage medium are provided, which are applicable to a field of computer technology, and particularly to a field of audio processing. The noise reduction method for an on-board environment includes: receiving an interference signal in the on-board environment and receiving a sound signal in the on-board environment, the interference signal comprising a vibration signal of a vehicle; and performing noise reduction processing on the sound signal in the on-board environment to obtain a noise-reduced signal; wherein, the noise reduction processing comprises cancelling the interference signal from the sound signal in the on-board environment.

Abstract: Systems and methods of providing improved directional noise suppression in an electronic device implement a technique that specifies a direction or speaker of interest, determines the directions corresponding to speakers not lying in the direction of interest, beam forms the reception pattern of the device microphone array to focus in the direction of interest and suppresses signals from the other directions, creating beam formed reception data. A spatial mask is generated as a function of direction relative to the direction of interest. The spatial mask emphasizes audio reception in the direction of interest and attenuates audio reception in the other directions. The beam formed reception data is then multiplied by the spatial mask to generate an audio signal with directional noise suppression.

Abstract: An Nth order linear differential microphone array (LDMA) with a steerable beamformer may be constructed by specifying a target beampattern for the LDMA at a steering angle ?. An Nth order polynomial associated with the target beampattern may then be generated. A relationship between the nulls of the polynomial and the steering angle ? is determined and then a value of one of the nulls is determined based on N?1 assigned values for the other nulls and the determined relationship between the nulls of the polynomial and the steering angle ?. The steerable beamformer may be generated based on the determined null value and the N?1 assigned null values. The N?1 assigned null values may be associated with the N?1 nulls of the polynomial that are of less than Nth order and the determined null value may be associated with the null of the polynomial that is of Nth order.

Abstract: A conversation support device includes: a speaker; a microphone; a noise source acquisition unit that acquires a noise signal indicating noise; a first calculator that calculates a transfer characteristic of a secondary path between the speaker and the microphone; an echo cancellation unit that cancels an echo by using the transfer characteristic of the secondary path; a second calculator that calculates a coefficient of an adaptive filter, based on the transfer characteristic of the secondary path and the noise signal; and an active noise cancellation controller that generates a noise cancelling signal by using the coefficient of the adaptive filter and the noise signal. The noise cancelling signal is for controlling cancellation of the noise.

Abstract: A speech enhancement method and a speech enhancement system are provided. The speech enhancement method performs two-stage noise suppression by using digital signal processing and neural network approach. The first-stage noise suppression generates artifact signals by reducing stationary noise in the digital audio signals. The second-stage noise suppression performs voice activity detection and further reduces non-stationary noise in the artifact signals. The result of the voice activity detection is fed back to establish or update a noise model used in the first-stage noise suppression.

Abstract: The present disclosure relates to circuitry comprising: amplifier circuitry configured to receive a variable supply voltage, wherein the supply voltage varies according to an output signal of the amplifier circuitry; monitoring circuitry configured to monitor one or more parameters of an output signal of the amplifier circuitry; and processing circuitry configured to receive an indication of the voltage of the variable supply voltage and an indication of the monitored parameters from the monitoring circuitry and to apply a correction to one or more of the monitored parameters to compensate for coupling between the variable supply voltage and the monitoring circuitry.

Abstract: Embodiments of wireless audio systems and methods for operation mode switch of wireless headphones are disclosed herein. In one example, a wireless audio system includes a first wireless headphone and a second wireless headphone. The first wireless headphone is configured to establish a first short-range wireless link with an audio source; transmit, to a second wireless headphone, link information associated with the first short-range wireless link; and remove the first short-range wireless link with the audio source in response to the second wireless headphone successfully establishing a second short-range wireless link with the audio source based on the link information. The second wireless headphone is configured to receive, from the first wireless headphone, the link information associated with the first short-range wireless link; and establish the second short-range wireless link with the audio source based on the link information.

Abstract: A method and device for detecting whether a headset is on ear. A probe signal is generated for acoustic playback from a speaker. A microphone signal from a microphone is received, the microphone signal comprising at least a portion of the probe signal as received at the microphone. The microphone signal is passed to a state estimator, to produce an estimate of at least one parameter of the portion of the probe signal contained in the microphone signal. The estimate of the at least one parameter is processed to determine whether the headset is on ear.

Abstract: A sound output device includes a speaker that outputs sound, a housing that includes a sound introduction hook, the sound introduction hook having an internal space that is formed as a sound introduction space for introducing sound output from the speaker, the speaker being arranged in the housing, and an ear pad portion that is provided to be continuous to the housing, the sound introduction hook including an opening through which sound is emitted toward an ear, the sound introduction hook being to be mounted on a root portion of an auricle from a helix side, the ear pad portion being to be mounted on the root portion of the auricle from an earlobe side.

Abstract: Provided are a method and apparatus for localizing a multichannel sound signal. The method includes: obtaining a multichannel sound signal to which sense of elevation is applied by applying a first filter to an input sound signal; determining at least one frequency range of a dynamic cue according to change of a head-related transfer function (HRTF) indicating information regarding paths from a spatial location of an actual speaker to ears of an audience; and applying a second filter to at least one sound signal, corresponding to the determined at least one frequency range, of at least one channel in the multichannel sound signal to change the at least one sound signal so as to remove or to reduce the dynamic cue when the multichannel sound signal is output.

Abstract: A vehicle sound synthesis system is provided with a controller and a loudspeaker. The controller is programmed to receive an input indicative of at least one of a gear selection, an engine speed, and a pedal position and to generate an audio signal indicative of synthesized engine noise (SEN). The controller is further programmed to attenuate the audio signal at a first rate in response to at least one of the gear selection, the engine speed, and the pedal position indicating first vehicle conditions. The loudspeaker is adapted to project sound within a passenger compartment of a vehicle in response to receiving the attenuated audio signal.

Abstract: Systems, methods, and computer program products for frequency-domain estimation of latency between audio signals. In some embodiments, the estimation is performed on first blocks of data indicative of samples of a first audio signal and second blocks of data indicative of samples of a second audio signal, and includes determining a coarse latency estimate, including by determining gains which, when applied to some of the second blocks, determine estimates of one of the first blocks, and identifying one of the estimates as having a best spectral match to said one of the first blocks. A refined latency estimate is determined from the coarse estimate and some of the gains. Optionally, at least one metric indicative of confidence in the refined latency estimate is generated. Audio processing (e.g., echo cancellation) may be performed on the frequency-domain data, including by performing time alignment based on the refined latency estimate.

Abstract: A new hearing aid is provided in which signals that are received from an external device, such as a spouse microphone, a media player, a hearing loop system, a teleconference system, a radio, a TV, a telephone, a device with an alarm, etc., are filtered in such a way that a user can localize the monaural signal transmitter.

Abstract: Embodiments included herein generally relate to using non-audio data embedded in an audio signal. More particularly, embodiments relate to using non-audio data embedded into the audio signal to control an audio configuration of a plurality of speakers and/or to measure a delay of a playback device.