Abstract: An audio apparatus includes: a receiver which receives audio data transmitted by a broadcast method from an external apparatus; a sound emitter which performs sound emission of the audio data received by the receiver; a generator which generates sound emission control information relating to sound emission of other apparatus which can receive the audio data; and a transmitter which transmits the sound emission control information generated by the generator to the other apparatus.

Abstract: A system for enabling a shared three-dimensional (“3D”) audio bed available to multiple software applications is provided. The system manages bed metadata defining a number of speaker objects of a 3D audio bed. The bed metadata also associates each speaker object with a location, which in some configurations, is defined by a three-dimensional coordinate system. The bed metadata is communicated to a plurality of applications. The applications can then generate custom 3D audio data that associates individual audio streams with individual speaker objects of the 3D audio bed. The applications can then communicate the custom 3D audio data to a 3D audio bed engine, which causes the processing and rendering of the custom 3D audio data to an output device utilizing a selected spatialization technology. Aspects of the 3D bed can be altered when the spatialization technology or the output device changes.

Abstract: An audio signal processing decoder having at least one frequency band and being configured for processing an input audio signal having a plurality of input channels in the at least one frequency band, wherein the decoder is configured to analyze the input audio signal, wherein inter-channel dependencies between the input channels are identified; and to align the phases of the input channels based on the identified inter-channel dependencies, wherein the phases of input channels are the more aligned with respect to each other the higher their inter-channel dependency is; and to downmix the aligned input audio signal to an output audio signal having a lesser number of output channels than the number of the input channels.

Abstract: A method for processing a signal includes receiving a first frame of an input audio bitstream at a decoder. The first frame includes at least one signal associated with a frequency range. The method also includes decoding the at least one signal to generate at least one decoded signal having an intermediate sampling rate. The intermediate sampling rate is based on coding information associated with the first frame. The method further includes generating a resampled signal based at least in part on the at least one decoded signal. The resampled signal has an output sampling rate of the decoder.

Abstract: The methods described herein are configured to accurately propagate sound through a virtual environment. During runtime of an application using the virtual environment, a shortest path is calculated from a sound source in a first zone to a sound destination in a second zone that passes through at least one portal. The direction of the calculated shortest path from the portal to the sound destination is determined. Then, a virtual sound source is generated based on the length and determined direction of the calculated path. Further, obstructions in the virtual environment are processed to determine an attenuation of audio data associated with the sound source. The generated virtual sound source and the attenuated audio data are provided to an audio engine for rendering to a user. Propagating sound using a combination of these methods provides an accurate, realistic sound experience in the virtual environment.

Abstract: Techniques for noise suppression systems coupled to one or more microphone arrays are described, including a housing, a first microphone, a second microphone, and a third microphone, where the third microphone functions as a common rear vent for the first and the second microphones.

Abstract: A device for processing audio signals includes an interchannel temporal mismatch analyzer, an interchannel phase difference (IPD) mode selector and an IPD estimator. The interchannel temporal mismatch analyzer is configured to determine an interchannel temporal mismatch value indicative of a temporal misalignment between a first audio signal and a second audio signal. The IPD mode selector is configured to select an IPD mode based on at least the interchannel temporal mismatch value. The IPD estimator is configured to determine IPD values based on the first audio signal and the second audio signal. The IPD values have a resolution corresponding to the selected IPD mode.

Abstract: A coil assembly is coupled to a diaphragm. The coil assembly has a number of coils that are fixed in a stacked, end to end manner along a length axis. The coils include a middle coil, a several upper coils and several lower coils. Each coil has a respective pair of coil terminals so that it can be independently driven by an audio signal. The coil assembly as a whole is overhung, while each of constituent coils is underhung so that there are at least two adjacent coils that are completely within the magnetic gap, and at least two other coils that are completely outside the magnetic gap, during a majority of the full excursion of the coil assembly. Other embodiments are also described and claimed.

Abstract: A wireless speaker device having a housing including a first magnet and an arm configurable between a straight configuration and a bent configuration. The arm is attached to the housing on one end and has an opposing end having a second magnet. The arm is bendable to allow the magnets to engage another to attach the device to clothing and other items.

Abstract: Various arrangements for providing audio calibration are presented. A remote control may determine that it is in position for an audio-based command to be received by the remote control from a user. Via a condenser microphone incorporated as part of the remote control, a calibration sound may be output. The condenser microphone may receive the output the calibration sound in the form of a reflected sound that is an acoustic reflection of the calibration sound. The remote control may analyze an amount of time that has elapsed between outputting the calibration sound and receiving the reflected sound to determine a distance between the condenser microphone and the user. One or more audio input settings may be configured that are used for modifying audio received by the condenser microphone based on the determined distance.

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: 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: Methods and apparatuses for addressing open space noise are disclosed. In one example, a method for masking open space noise includes receiving a microphone output signal from a microphone, the microphone one of a plurality of microphones in an open space. The method includes detecting a presence of a noise source from the microphone output signal, and determining whether the noise source is capable of being masked with a noise masking sound. The method further includes increasing a volume of a noise masking sound output from a loudspeaker responsive to a determination that the noise source is capable of being masked, the loudspeaker located in a same geographic sub-unit of the open space as the microphone. The loudspeaker is one of a plurality of loudspeakers in the open space.

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: Systems and techniques for providing improved beam forming functionality and/or improve noise cancellation functionality in a microphone package are presented. In an implementation, a device includes a first microelectromechanical systems (MEMS) sensor, a second MEMS sensor, and a digital signal processor. The first MEMS sensor is contained in a first back cavity within the device. The second MEMS sensor is contained in a second back cavity within the device. The digital signal processor generates a cardioid microphone pattern based on data associated with the first MEMS sensor contained in the first back cavity and other data associated with the second MEMS sensor contained in the second back cavity, where the digital signal processor forms at least a portion of the first back cavity and the second back cavity.

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 terminal device includes: a first housing including a first opening and a second opening through which sound of a speaker is output; a second housing which is turnably coupled to the first housing and closes the first opening based on a turn angle relative to the first housing; and a closing member which closes the second opening in a state where the first opening is opened by the second housing.

Abstract: A computer-implemented system and method for identifying and locating an emitted acoustic signal from a source entity is disclosed. The system and method is capable of distinguishing between source entities of the same type. The system and method may further allow a user of the system to observe the movement of a source entity in real-time from a remote location as well as access stored data representative of the movement of the source entity from a remote location. The system and method is capable of identifying the identity of a source entity by using an average sound source comparing a received emitted acoustic signal from the source entity to a database of sounds. The system and method is capable calculating the source entity's distance and direction from an origin input sensor for each occurrence of emitted acoustic signals received from the source entity.

Abstract: Techniques are provided for QR Decomposition (QRD) based minimum variance distortionless response (MVDR) adaptive beamforming. A methodology implementing the techniques according to an embodiment includes receiving signals from microphone array, identifying signal segments that include a combination of speech and noise, and identifying signal segments that include noise in the absence of speech. The method also includes calculating a QRD and an inverse QRD (IQRD) of the spatial covariance of the noise components. The method further includes estimating a relative transfer function (RTF) associated with the source of the speech, based on the noisy speech signal segments, the QRD, and the IQRD. The method further includes estimating a multichannel speech-presence-probability (SPP) on whitened input signals based on the IQRD. The method further includes calculating beamforming weights, for the microphone array, based on the RTF and the IQRD, to steer a beam in the direction associated with the speech source.

Abstract: A pneumatic audio system which is compatible with use within an MRI environment is disclosed. The system has an audio transducer including a housing having a plurality of walls, a foam insert supported within the housing, and a speaker supported by the foam insert and extending at an angle with respect to the walls of the housing. The system further has a hollow tube coupled to the audio transducer at a first end and configured to transmit audio signals from the speaker through the tube. The system further has an earphone coupled to a second end of the hollow tube and configured to deliver audio signals to an ear canal of a human user.