Abstract: Audio mix information is received from a plurality of users. Mix rules are determined from the audio mix information from the plurality of users, wherein the mix rules include a first mix rule associated with a first audio item. The first mix rule relates to an overlap of the first audio item with another audio item. The first mix rule is made available to one or more clients. After making the first mix rule available, an indication, from a respective client device, that the first audio item is to be mixed with a second audio item at the respective client device in accordance with the first mix rule is received. In response to the indication, a specification of the first mix rule is transmitted to the respective client device to be applied by the respective client device to generate a transition between the first audio item and the second item.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: Described herein are systems, devices, and methods for discovering and assigning wireless speakers to one or more devices capable of providing audio content. In some embodiments, the speakers may receive one or more channels of multi-channel audio content. In some embodiments, one or more wireless speakers are discovered, a prompt identifying at least one speaker location is provided at an output device, a list of wireless speakers is provided to the output device, wherein the list of wireless speakers corresponds to the discovered one or more wireless speakers, a selection of a wireless speaker from the list of wireless speakers is received; and the selected wireless speaker is associated to the at least one speaker location.

Abstract: A system includes a class D amplifier and a current steering digital-to-analog converter (DAC) directly connected to the class D amplifier. The system also includes a common mode servo circuit coupled to a node interconnecting the current steering DAC to the class D amplifier. The common servo circuit amplifies a difference between a common mode signal determined from the node and a reference voltage and generates a feedback current to the node based on the amplified difference. A feed-forward common-mode compensation circuit is included to reduce an alternating current (AC) ripple from the class D amplifier. The feed-forward common-mode compensation circuit includes first and second resistors coupled to respective outputs of the class D amplifier. A current mirror is coupled to the first and second resistors and is configured to sink a current from the node to ground that approximates a common mode feedback current of the class D amplifier.

Abstract: Audio systems and methods are provided to increase relevant content of an audio signal relative to a noise level, and include filtering the audio signal to remove spectral content below a cutoff frequency, where the cutoff frequency is based upon a noise value.

Abstract: Examples described herein involve calibrating a playback device. An example implementation receives, from a network microphone device (NMD), data indicating second audio signal detected by the NMD at multiple locations between a first physical location and a second physical location within a given environment while the network microphone device is moving from the first physical location to the second physical location, the second audio signal representing acoustic echo of a first audio signal played by a playback device. Based on the detected second audio signal, the implementation determines an audio characteristic of the given environment. Based on the determined audio characteristic, the implementation determines an audio processing algorithm to adjust audio output of the playback device in the given environment to have a pre-determined audio characteristic that is representative of desired audio playback qualities.

Abstract: A sports Hi-Fi touch-control Bluetooth earphone includes a neckband, a first case and a second case connected to both ends of the neckband, an earplug set on each of the first and second cases, a Bluetooth main circuit board, a battery, a main microphone, an auxiliary microphone, and a capacitive touchpad. The main microphone and the auxiliary microphone simultaneously receive audio signals. The differential amplifier of the Bluetooth main circuit board computes the signal difference between the main microphone and the auxiliary microphone. The signal difference is further processed by a band-pass filter to pass only the human voice frequency band, with low and high frequency signals filtered out. The human voices are amplified to produce clear conversations, enhancing conversation quality and avoiding unclear signals. Mechanical keys on conventional Bluetooth earphones are replaced by the capacitive touchpad, making operations easy, convenient, waterproof, and dustproof to extend the lifetime thereof.

Abstract: A system with a plurality of microphones positioned at different locations, and a modification system in communication with the microphones. The modification system is configured to derive a plurality of audio signals from the plurality of microphones, compute a confidence score for each derived audio signal, and based on the computed confidence scores, use one derived audio signal to modify another audio signal.

Abstract: The present disclosure provides a number of techniques for personalization of audio for communication to an endpoint device. According to one technique, a cloud-based computing device may receive media data from a media platform, and generate depersonalized audio data based on the media data, the depersonalized audio data including at least one audio component associated with the media platform. The technique may further generate user personalized audio data based on the depersonalized audio data. The user personalized audio data may include at least one audio component personalized based on a unique head-related transfer function (HRTF) data associated with a user or default HRTF data. The user personalized audio may be communicated to the endpoint device for consumption by the user. Such techniques reduce the need for advanced audio processing technologies on client systems and makes personalized audio more available to users.

Abstract: Systems, methods, and non-transitory computer-readable storage media for performing a role swapping operation between a pair of non-tethered wireless ear buds after detecting a triggering event. Further, state information can be coordinated between devices, including in connection with performing a role swap between buds in a pair of wireless, untethered ear buds, where one wireless ear bud is in a primary role and is responsible for a connection with a companion device, and another wireless ear bud in the pair is in a secondary role.

Abstract: This document discusses, among other things, systems and methods to reduce power use of an accessory detection device. The accessory detection device can be configured to be coupled to a mobile device having an audio jack configured to be coupled to a mobile device accessory having a send/end key. In an example, the accessory detection device can include a comparator and a switch. The comparator can be configured to receive mobile device accessory information from the mobile device accessory and to determine activation of the send/end key using the received mobile device accessory information. The switch can be configured to receive connection information indicative of mobile device accessory connection to the audio jack and to isolate a reference input of the comparator from a supply voltage using the connection information, for example, to reduce leakage current.

Abstract: An apparatus includes a sensor module configured for receiving sensed information indicative of a sensed signal. The sensed signal includes a source signal component and a source noise component. The apparatus also includes a reference module configured for reference information indicative of a reference signal. The reference signal also includes a reference noise component. The apparatus also includes a filter module configured as a fixed lag Kalman smoother. The filter module is configured for adaptively filtering the reference signal to generate an estimate of the source noise component. The apparatus also includes a processing module configured for calculating an output signal based on the sensed signal and the estimate of the source noise component. The apparatus also includes an interface module configured for transmitting an indication of the output signal. The filter module is further configured for, based on the output signal, tuning the Kalman smoother.

Abstract: In one aspect, the invention divides an audio signal into auditory events, each of which tends to be perceived as separate and distinct, by calculating the spectral content of successive time blocks of the audio signal, calculating the difference in spectral content between successive time blocks of the audio signal, and identifying an auditory event boundary as the boundary between successive time blocks when the difference in the spectral content between such successive time blocks exceeds a threshold. In another aspect, the invention generates a reduced-information representation of an audio signal by dividing an audio signal into auditory events, each of which tends to be perceived as separate and distinct, and formatting and storing information relating to the auditory events. Optionally, the invention may also assign a characteristic to one or more of the auditory events. Auditory events may be determined according to the first aspect of the invention or by another method.

Abstract: At a microphone array, a soundfield is detected to produce a set of microphone signals each from a corresponding microphone in the microphone array. The set of microphone signals represents the soundfield. The detected soundfield is decomposed into a set of sub-soundfield signals based on the set of microphone signals. Each sub-soundfield signal is processed, such that each sub-soundfield signal is separately dereverberated using other ones of the sub-soundfield signals to remove reverberation from the sub-soundfield signal, to produce a set of processed sub-soundfield signals. The set of processed sub-sound field signals are mixed into a mixed output signal.

Abstract: A method, and a corresponding apparatus, for processing an input signal comprise filtering the input signal to separate a passband frequency component of the input signal from a stopband frequency component of the input signal, and adjusting relative signal power values of the passband frequency component and the stopband frequency component of the input signal based at least in part on signal values of a number of samples associated with the input signal. In the case of audio signals, for example, such processing is used for spectral expansion of the input signal by enhancing the power of the stopband, or low and high frequencies, component with respect to the power of the passband component of the input signal. As a result, a better audio quality is achieved.

Abstract: An audio system comprises an audio driver configured to receive a target audio signal and a feedback signal and to generate an adjusted audio signal responsive to the target audio signal and the feedback signal. A loudspeaker is configured to convert the adjusted audio signal into acoustical sound. A test signal generator is configured to generate a test signal having a higher frequency than the target audio signal. The test signal causes a test current to flow through the loudspeaker. A current sensing circuit is configured to measure the test current flowing through the loudspeaker and to generate a current sense signal indicative of the test current. A feedback circuit is configured generates the feedback signal responsive to the current sense signal.

Abstract: A system for tuning an audio reproduction device is disclosed. The system includes an audio signal generator for generating an audio signal to be played back in an audio reproduction device and sending the audio signal to the audio reproduction device; a microphone signal module for receiving a microphone signal from a microphone, the microphone signal recording a sound wave reproduced by the audio reproduction device when the audio signal is played back in the audio reproduction device; a comparison module for determining whether the microphone signal matches target sound signature data; and a filter module for applying a digital filter to emulate a target sound signature responsive to determining that the microphone signal does not match the target sound signature data.

Abstract: A method for characterizing a loudspeaker comprises: filtering an input test signal by applying the inverse of a linear model describing an excursion of the loudspeaker in response to a given input signal, the linear model containing only linear terms; applying the filtered input test signal to the loudspeaker; measuring the excursion of the loudspeaker in response to the filtered input test signal; and, based on the measured excursion, determining one or more non-linear parameters for a non-linear model describing the excursion of the loudspeaker in response to a given input signal.

Abstract: A portable loudspeaker includes an enclosure that defines an acoustic cavity. An electro-acoustic transducer is disposed within the acoustic cavity. A strap is provided which has a first end that is fixedly coupled to the enclosure and a second end that is releasably coupled to the enclosure. The second end of the strap is configured to be coupled to the enclosure (e.g., to secure an object therebetween). The second end of the strap is also configured be coupled to another portion of the strap to form a closed loop.

Abstract: A device includes a processor, a memory, and a combiner. The processor is configured to receive a first combined frame and a second combined frame corresponding to a multi-channel audio signal. The memory is configured to store first lookahead portion data of the first combined frame. The first lookahead portion data is received from the processor. The combiner is configured to generate a frame at a multi-channel encoder. The frame includes a subset of samples of the first lookahead portion data, one or more samples of updated sample data corresponding to the first combined frame, and a group of samples of second combined frame data corresponding to the second combined frame.