Abstract: A system for detection of a target sound in an environment of a vehicle, includes an audio sensor, a computer processor, and a memory storing digital target sound templates produced by converting samples of the target sound in accordance with conversion parameters. The computer processor receives the sample sound signal from the audio sensor, converts the sample sound signal to a digital sample vector in accordance with the conversion parameters, compares the vector to each of the templates to determine if a degree of similarity of the vector to any one of the templates exceeds a predetermined threshold value, and notifies a user or a vehicle control system if the threshold value is exceeded.

Abstract: An embodiments of a contextual sound apparatus may include a sound identifier to identify a sound, a context identifier to identify a context, and an action identifier communicatively coupled to the sound identifier and the context identifier to identify an action based on the identified sound and the identified context. Other embodiments are disclosed and claimed.

Abstract: An electronic device includes a memory, a communication interface capable of performing communication, and a processor electrically connected to the memory and the communication interface. The processor is configured to extract a first audio signal to be output through a speaker, based on audio data received from an external electronic device and/or stored in the memory, to generate a second audio signal obtained by removing the first audio signal from the audio data, and to generate a haptic signal for outputting a haptic effect, based on the second audio signal.

Abstract: One or more accelerometers embedded with an earbud and/or a set of earphones are able to sense a moving pace of a user. Based on a moving pace of the user, a signal is sent to a remotely connected electronic device. The electronic device is able to separately increase and decrease a beat or rhythm of the audio from the electronic device based on a pace of the user. In some embodiments, an audio alert is sent to the user to inform the user of pace and whether the user has increased or decreased their pace. Additionally, in some embodiments, a program stored on the electronic device is used to compare the user's current progress and/or speed based on past runs and workouts.

Abstract: The present application relates to packet loss concealment apparatus and method, and audio processing system. According to an embodiment, the packet loss concealment apparatus is provided for concealing packet losses in a stream of audio packets, each audio packet comprising at least one audio frame in transmission format comprising at least one monaural component and at least one spatial component. The packet loss concealment apparatus may comprises a first concealment unit for creating the at least one monaural component for a lost frame in a lost packet and a second concealment unit for creating the at least one spatial component for the lost frame. According to the embodiment, spatial artifacts such as incorrect angle and diffuseness may be avoided as far as possible in PLC for multi-channel spatial or sound field encoded audio signals.

Abstract: A dual microphone near field voice enhancement arrangement in a motor vehicle includes a seat having a headrest with two opposite lateral sides. Each of two microphones is mounted on a respective one of the two opposite lateral sides of the headrest. Each microphone produces a respective microphone signal indicative of sounds within a passenger compartment. An electronic processor receives the microphone signals. The processor calculates a time delay between the microphone signals, and uses the calculated time delay to estimate amplitudes of the microphone signals. The processor applies a respective time delay to each of the microphone signals based on the calculated time delay to produce two time-aligned signals. The processor applies a respective gain to each of the time-aligned microphone signals based on the estimated amplitudes to produce two time-aligned and gain corrected signals. The processor sums the time-aligned and gain corrected signals.

Abstract: A method includes generating a windowed time-domain mid channel by applying two first asymmetric windows to a first frame of a time-domain mid channel and applying two second asymmetric windows to a second frame of the time-domain mid channel. The method includes transforming the windowed time-domain mid channel to a transform domain to generate sets of transform-domain mid channel data including first transform-domain mid channel data corresponding to a first mid channel window of the first frame and second transform-domain mid channel data corresponding to a second mid channel window of the first frame. The method includes performing an up-mix operation using the sets of transform-domain mid channel data, stereo parameters from the bit stream, and an interpolated parameter determined using an unevenly weighted interpolation between a first stereo parameter value associated with the first frame and a second stereo parameter value associated with the second frame.

Abstract: The method and apparatus described herein make use of multiple sets of head related transfer functions (HRTFs) that have been synthesized or measured at various distances from a reference head, spanning from the near-field to the boundary of the far-field. Additional synthetic or measured transfer functions maybe used to extend to the interior of the head, i.e., for distances closer than near-field. In addition, the relative distance-related gains of each set of HRTFs are normalized to the far-field HRTF gains.

Abstract: Embodiments herein relate generally to changing spatial audio fields that are defined for audio sources. In the embodiments, the spatial audio fields are indicated to a user performing audio mixing, for instance by displaying them as polygons on a touch screen. The spatial audio fields move as the related audio sources move, and/or as the position of a notional consumer changes. Apparatus of the embodiments is configured to detect whether at any time (initially, or after movement) there is overlapping of two spatial audio fields. If an overlap is detected, this is indicated to a user performing audio mixing The apparatus then responds to a user input (e.g. a gesture on the touch screen) by detecting the nature of the user input and then moving or sizing one or both of overlapping spatial audio fields and such that overlapping is avoided or reduced.

Abstract: According to one or more aspects of the present disclosure, operations may include obtaining multiple microphone signals derived from a microphone array that includes a multiple omnidirectional microphones. Each of the microphone signals may be derived from a different microphone of the microphone array. The operations may further include determining whether the microphone signals include noise, such as wind noise, based on two or more of the plurality of microphone signals. In addition, the operations may include generating an output signal based on a beamformed signal or a reduced-noise signal based on whether the microphone signals are determined to include noise.

Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component ({tilde over (C)}AMB(k?1)). The ambient HOA component ({tilde over (C)}AMB(k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (cAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.

Abstract: A system and method for quieting unwanted sound. As a non-limiting example, various aspects of this disclosure provide a system and method, for example implemented in a premises-based or home audio system, for quieting unwanted sound at a particular location.

Abstract: An integrated circuit arrangement for a microphone, a microphone system and a method for adjusting circuit parameters of the microphone are disclosed. In an embodiment an integrated circuit includes an amplifier circuit with a first switchable network circuit for adjusting an amplifier current of the amplifier circuit, the first switchable network circuit comprising a plurality of switches (SW1, . . . , SWx) each coupled with a first control port of the first switchable network circuit and a control unit coupled with the first control port of the first switchable network circuit and configured to control a setting of the respective switches (SW1, . . . , SWx) of the first switchable network circuit.

Abstract: A device is provided which adds a desired delay (echo) effect while suppressing increase in the cost with a simple structure. A CPU of an audio device activates a plurality of media players A, B, C, . . . for reproducing an audio file. The audio file is reproduced as an original sound by one of the plurality of media players, and a delay sound with respect to the original sound of the audio file is reproduced by another media player, and is added to the original sound.

Abstract: A method includes receiving sound by a first audio unit installed in an electrical outlet, routing audio data corresponding to the received sound from the first audio unit to a second audio unit installed in a second electrical outlet, and sending the audio data to a mobile device using a wireless link between the mobile device and the second audio unit. Routing the audio data may include receiving the audio data from the first audio unit by a third audio unit and routing the audio data to the second audio unit by the third audio unit serving as a router. The data may be routed using table driven routing, on-demand routing or some other appropriate routing protocol. The method may also include performing voice recognition on the audio data and detecting a command word and routing command word data to the second audio unit.

Abstract: Systems and techniques for compression and decoding of audio data are generally disclosed. An example device for compressing higher order ambisonic (HOA) coefficients representative of a soundfield includes a memory configured to store audio data and one or more processors configured to: determine when to use ambient HOA coefficients of the HOA coefficients to augment one or more foreground audio objects obtained through decomposition of the HOA coefficients based on one or more singular values also obtained through the decomposition of the HOA coefficients, the ambient HOA coefficients representative of an ambient component of the soundfield.

Abstract: A MEMS device includes a backplate electrode and a membrane disposed spaced apart from the backplate electrode. The membrane includes a displaceable portion and a fixed portion. The backplate electrode and the membrane are arranged such that an overlapping area of the fixed portion of the membrane with the backplate electrode is less than maximum overlapping.

Abstract: An improved Class-D amplifier may include a digital PWM modulator that receives a digital audio signal and that provides a PWM signal as an input to an analog PWM modulator of the Class-D amplifier. A digital PWM modulator may be configured to receive a digital signal and to generate a first PWM signal based on the digital signal. A digital PWM driver may be coupled to the digital PWM modulator. The digital PWM driver may be configured to receive the first PWM signal and to generate a reference PWM signal based on the first PWM signal. An analog PWM modulator may be coupled to the digital PWM driver. The analog PWM modulator may be configured to receive the reference PWM signal and to generate an output PWM signal based on the reference PWM signal.

Abstract: Disclosed are a sound sharing apparatus and a sound sharing method. The sound sharing apparatus according to one embodiment of the present disclosure includes at least one processor configured to implement: a modifier configured to change a default audio render driver of a local machine from a first audio render driver to a second audio render driver; a capturer configured to capture audio data transmitted to the second audio render driver; and a mixer configured to mix the captured audio data: i) with first voice data to output first mixed data, wherein the first voice data is received from a remote machine connected to the local machine through a network, or ii) with second voice data to output second mixed data, wherein the second voice data is received input through a microphone of the local machine.

Abstract: A mixing console including a plurality of audio inputs and a plurality of audio processing channels. Control data is received from a microphone which is connected to a given audio input to provide audio data. The control data includes an indication of an audio source associated with a microphone. A router of the mixing console is configured to route the audio data from the given audio input to a given audio processing channel based on the received indication of the audio source.