Abstract: A system for managing moderator and audience audio captured by personal mobile devices and broadcast by a public address system in the same venue is disclosed. In various aspects, the system includes a wireless network hub. The wireless network hub transmits and receives wireless data from a plurality of personal mobile devices. A network manager component manages communications of the personal mobile devices with the wireless network hub. The network manager uses measured dynamic network parameters and determines conditions for maintaining voice stream quality according to configurable rules.

Abstract: The present invention is a modular signal converting apparatus and method, and particularly, discloses a signal converting apparatus, which is modularized for playback of digital contents and is usable while being combined with another electric device.

Abstract: An in-ear headphone includes a shell, a speaker, a circuit board, and a sensing structure. The shell has an accommodating space. The speaker is disposed in the accommodating space. The circuit board is disposed in the accommodating space, and the circuit board is electrically connected to the speaker. A sensing chip is disposed on the circuit board. The sensing structure includes a sensing sound tube and a transmission member. The sensing sound tube protrudes from a front end of the shell and has a conductor. The sensing sound tube is electrically connected to the sensing chip by the transmission member. A material of the sensing sound tube is different from a material of the shell.

Abstract: In general, various aspects of the techniques are described for interpolating audio streams. A device comprising a memory and a processor may be configured to perform the techniques. The memory may store the one or more audio streams. The processor may obtain one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams. The processor may also obtain a listener location identifying a location of a listener, and perform interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream. The processor may next obtain, based on the interpolated audio stream, one or more speaker feeds, and output the one or more speaker feeds.

Abstract: Embodiments include systems with active sound canceling properties, fenestration units with active sound canceling properties, retrofit units with active sound canceling properties and related methods. In an embodiment a system can include a sound cancellation device include a sensing element to detect vibration of a transparent pane and/or a sound input device configured to detect sound incident on the transparent pane, as well as a vibration generator configured to vibrate the transparent pane and a sound cancellation control module. The sound cancellation control module can evaluate the detected vibration of the transparent pane at two or more discrete frequency bands. The sound cancellation control module can cause the vibration generator to vibrate the transparent pane causing destructive interference with sound waves at the two or more discrete frequency bands. Other embodiments are also included herein.

Abstract: A sound source separation method, applied in a sound system, is provided. The method comprises choosing a maximum sound source signal and at least a non-maximum sound source signal from a plurality of sound source signals; multiplying the at least a non-maximum sound source signal by at least a suppression value, to generate at least a suppressed sound source signal; and performing a back-end sound source extraction operation on the maximum sound source signal and the at least a suppressed sound source signal.

Abstract: An interactive music tag system and an interactive method thereof are provided in the present invention. The interactive method is adapted for an interactive music tag and an interactive music tag reader. The method includes: when the interactive music tag is close to the interactive music tag reader, outputting an wireless energy to the interactive music tag and outputting an audio envelop signal to the coil of an LC resonant circuit such that the analog audio signal is transmitted to the interactive music tag; when the interactive music tag reader controls the interactive music tag, modulating a control signal into the wireless energy; when the interactive music tag receives the control signal through the LC resonant circuit, controlling the output status of the audio envelop signal according to the control signal.

Abstract: A head-mounted wearable device (HMWD) provides audio output from a first speaker that is driven with a first signal and a second speaker that is driven with a second signal. Based on a volume level setting, an equalization profile and inversion frequency are determined. The equalization profile selectively amplifies or attenuates particular frequencies or ranges of frequencies. Those frequencies in the second signal that are above the inversion frequency have their amplitude inverted, relative to the first signal. When driven by the first signal and the second signal, the first speaker and the second speaker operate as acoustic dipoles below the inversion frequency and acoustic quadrupoles above the inversion frequency. Sound from the first and second speakers with frequencies above the inversion frequency exhibits destructive interference. As a result, the user wearing the HMWD is able hear audio output while audio amplitude perceived by the bystanders is significantly reduced.

Abstract: Embodiments of source separation for reverberant environment are disclosed. According to a method, first microphone signals for each individual one of at least one source are captured respectively by at least two microphones for a period during which only the individual one produces sounds. Mixing parameters for modeling acoustic paths between the at least one source and the at least two microphones are learned by a processor based on the first microphone signals. Second microphone signals are captured respectively by the at least two microphones for a period during which all of the at least one source produce sounds. The reconstruction model is estimated by the processor based on the mixing parameters and second microphone signals. The processor performs the source separation by applying the reconstruction model.

Abstract: A active noise cancellation (ANC) system may include an adaptive filter divergence detector for detecting divergence of the one or more controllable filters as they adapt, based on various temporal or frequency domain amplitude characteristics. Upon detection of a controllable filter divergence, the ANC system may be deactivated, or certain speakers may be muted. Alternatively, the ANC system may modify the diverged controllable filters to restore proper operation of the noise cancelling system. This may include adjusting a leakage value of an adaptive filter controller.

Abstract: Systems and methods for automatically adjusting device volume based on learned volume preferences are disclosed herein. A first device receives a wireless signal from a second device. A signal strength of the wireless signal is determined, and a location of the second device is determined based on the signal strength of the wireless signal. Historical volume level data for the first device is retrieved from memory. A target volume level for the first device is determined based on the location of the second device and the historical volume level data. A volume setting of the first device is automatically adjusted to the target volume level.

Abstract: Embodiments are directed to an upward-firing speaker that reflects sound off a ceiling to a listening location at a distance from a speaker. The reflected sound provides height cues to reproduce audio objects that have overhead audio components. The speaker comprises a direct-firing tweeter and an upward-firing full-range driver in a unitary enclosure for playback of front-channel and height-channel signals, respectively. A crossover passes high frequencies of a front-channel signal directly to the tweeter and combines low frequencies of the front-channel signal with the height channel signal to be played through the full-range driver. A virtual height filter is applied to the height channel signal to improve the perception of height for audio signals transmitted by the virtual height speaker to provide optimum reproduction of the overhead reflected sound.

Abstract: A quantizer and a method for a sigma-delta modulator circuit that may be used as a component within an adaptive-noise cancelling headphone are presented. An apparatus includes a quantizer to receive an input signal with successive input values and quantizes the input signal at discrete intervals. This is done by mapping the input value of the input signal at each interval to one of a plurality of quantization levels with three or more quantization levels that are non-uniformly spaced. The plurality of quantization levels has a first portion with two or more quantization levels having the same sign and being proportional to a first fraction having one as its numerator and two to a power of a first variable as its denominator, the first variable being an integer and having a different value for each of the two or more quantization levels of the first portion.

Abstract: Techniques of performing linear acoustic echo cancellation performing a phase correction operation on the estimate of the echo signal based on a clock drift between a capture of an input microphone signal and a playout of a loudspeaker signal. Along these lines, the existence of the clock drift, i.e., a small difference in the sampling rates of the input microphone signal and the loudspeaker signal, can cause processing circuitry in a device configured to perform LAEC operations to generate a filter based on the magnitudes of the short-term Fourier transforms (STFTs) of the input microphone signal and the loudspeaker signal. Such a filter is real-valued and results in a positive estimate of the acoustic echo signal included in the input microphone signal. The phase of this estimate may then be aligned with the phase of the input microphone signal.

Abstract: A multichannel microphone-based reverberation time estimation method and device which use a deep neural network (DNN) are disclosed. A multichannel microphone-based reverberation time estimation method using a DNN, according to one embodiment, comprises the steps of: receiving a voice signal through a multichannel microphone; deriving a feature vector including spatial information by using the inputted voice signal; and estimating the degree of reverberation by applying the feature vector to the DNN.

Abstract: A method plays sound clips in binaural sound to users during an electronic communication between two or more users. A processor convolves the sound clips with HRTFs so the sound externally localize away from a head of the user listening to the sound clip. The sound clips play to users during the electronic communication without transmitting the sound clips to electronic devices of the users.

Abstract: Playing back of a piece of multimedia sound content by means of the vibration of any radiating structure, produced by contact with a playback device comprising: a mechanical actuator applying vibratory excitation to the structure, from a source signal, corresponding to a multimedia sound signal, a sensor sensing a signal from the vibration of the radiating structure, in order to thus: supply the actuator with a source signal, of known frequency components, receive, from the sensor, a vibration signal from the radiating structure operated by the actuator, in order to obtain a frequency response from the radiating structure, correct the source signal supplied to the actuator, by applying a filter correcting the spectral response of the radiating structure such that this response, once corrected, has a flat spectral profile, with amplitude deviations at most of 5 dB, and supply the actuator with the source signal corrected in this way.

Abstract: Embodiments of active noise control (ANC) headphones and operating methods thereof are disclosed herein. In one example, a headphone for ANC includes a speaker, an internal microphone, a processor, and a filter function module. The speaker is configured to play an audio based on a first audio source signal. The internal microphone is configured to obtain a mixed audio signal comprising a noise signal and a second audio source signal based on the audio of interest played by the speaker. The processor is configured to determine a current system parameter of the ANC headphone based on the mixed audio signal at a first time point and determine a current parameter of a filter function module based on the current system parameter of the ANC headphone and pre-tested data. The filter function module is to perform ANC based on the determined current parameter of the filter function module.

Abstract: Methods and systems for designing binaural room impulse responses (BRIRs) for use in headphone virtualizers, and methods and systems for generating a binaural signal in response to a set of channels of a multi-channel audio signal, including by applying a BRIR to each channel of the set, thereby generating filtered signals, and combining the filtered signals to generate the binaural signal, where each BRIR has been designed in accordance with an embodiment of the design method. Other aspects are audio processing units configured to perform any embodiment of the inventive method. In accordance with some embodiments, BRIR design is formulated as a numerical optimization problem based on a simulation model (which generates candidate BRIRs) and at least one objective function (which evaluates each candidate BRIR), and includes identification of a best one of the candidate BRIRs as indicated by performance metrics determined for the candidate BRIRs by each objective function.

Abstract: A system includes a logic device configured to cause the system to perform operations. The operations include requesting, from a peripheral device coupled to a host device, a device descriptor of the peripheral device, receiving, from the peripheral device, the device descriptor including at least two configurations. A first configuration includes an audio communications protocol having a plurality of selectable audio bit-rates and a second configuration including a plurality of selectable audio bit-rates and burst-rates. The operations further include selecting, from the first configuration and the second configuration based, at least in part, on a configuration of the host device, wherein the host device is configured to select the second configuration when the host device is compatible with the second configuration and select the first configuration when the host device is not compatible with the second configuration.