Abstract: Generating an audio signal with a configurable distance cue includes the following: receiving an input audio signal that has at least one signal characteristic that contributes to a listener's auditory distance perception; receiving a distance control signal; and changing at least one characteristic in accordance with a distance control signal that is representative of the perceived distance.

Abstract: An inspection signal generator of an acoustic device outputs a first inspection signal. An amplifier receives a first reference signal, and outputs a second inspection signal according to the first inspection signal. A first detector is electrically connected to an output side of the amplifier, and outputs a first detection signal based on the second inspection signal. The first detector is electrically connected to the output side of the amplifier, and outputs the first detection signal based on the second inspection signal. A determiner compares a first detection signal output from the first detector with a preset determination reference signal so as to detect electrical abnormality between a ground terminal and an output terminal electrically connected to the output side of the amplifier.

Abstract: In a one-to-many audio wireless transmission system in which a plurality of speaker devices on an audio reception side are provided with respect to one source device, in a case where an error has occurred in a certain speaker device, a possibility of the same error occurring in other speaker devices is reduced. A speaker device (2a, 2b) includes an error detection unit (23) which detects an error regarding sound output and transmits error information which is information regarding an error detected by the error detection unit (23) to a source device (1) by using wireless communication. The source device (1) transmits an operation request corresponding to the error information by using wireless communication to speaker devices other than the speaker device which has transmitted the error information.

Abstract: Apparatus for linear and nonlinear acoustic echo control includes loudspeaker, first, second, and third microphone, beamformer, and first echo canceller. The loudspeaker outputs a loudspeaker signal that includes reference signal. The first microphone and the second microphone are collocated with the loudspeaker, receive at least one of: a near-end speaker signal from a near-end speaker and the loudspeaker signal, and generate first and second microphone uplink signals, respectively. The third microphone receives the near-end speaker signal and generates a third microphone uplink signal. The beamformer receives the first and second microphone uplink signals, directs a beam towards the loudspeaker and drives a null towards the near-end speaker, and generates a beamformer output. The first echo canceler receives the third microphone uplink signal and the beamformer output, and cancels echoes in the third microphone uplink signal based on the beamformer output to generate an echo cancelled signal.

Abstract: An electronic device may include one or more microphones that monitor ambient noise and generate ambient noise data. The electronic device may include processing circuitry that receives the ambient noise data and compares the ambient noise data to a baseline ambient noise profile to detect changes in the ambient noise. Changes in the ambient noise may be caused by user gestures. The electronic device may compare the changes in the ambient noise to characteristic changes in ambient noise that are associated with user gestures to determine the user gesture made by a user. The processing circuitry may modify the operation of the electronic device based on the user gesture. An ambient noise input device may include surface features or movable components that cause the characteristic changes in the ambient noise. Ambient noise input devices may provide feedback to guide a user to provide input using ambient noise sensing.

Abstract: An audio spatial rendering apparatus and method are disclosed. In one embodiment, The audio spatial rendering apparatus includes a rendering unit for spatially rendering an audio stream so that the reproduced far-end sound is perceived by a listener as originating from at least one virtual spatial position, a real position obtaining unit for obtaining a real spatial position of a real sound source, a comparator for comparing the real spatial position with the at least one virtual spatial position; and an adjusting unit for, where the real spatial position is within a predetermined range around at least one virtual spatial position, or vice versa, adjusting the parameters of the rendering unit so that the at least one virtual spatial position is changed.

Abstract: Active vibration isolation (AVI) systems are becoming more available in various markets, one such market being vehicle operator seating. Unfamiliarity with the performance of such systems may cause users initial perceptions of system performance to be unfavorable. AVI systems can include a demonstration system capable of providing simulations too users of various types of vibration isolation systems under various input conditions, so that users can be introduced to the system benefits over other systems before using an AVI equipped product in its intended application.

Abstract: To input the voice of the passenger of a vehicle and output a piece of high-quality enhanced speech independently of the direction of a piece of speech or noise, a speech processing apparatus includes a first microphone that is provided on one of a ceiling member in a vehicle and an accessory thereof, inputs a sound mixture including a voice of a passenger of the vehicle and noise in the vehicle, and outputs a first signal, a second microphone that is provided on one of the ceiling member in the vehicle and the accessory thereof at a position farther than the first microphone when viewed from the passenger of the vehicle, inputs the noise in the vehicle while insulating the voice of the passenger of the vehicle using one of the ceiling member of the vehicle and the accessory thereof, and outputs a second signal, and a noise suppressor that outputs an enhanced speech signal based on the first signal and the second signal.

Abstract: [Object] To make it possible to further improve acoustic characteristics. [Solution] There is provided a headphone including: a driver unit including a vibration plate; a housing configured to house the driver unit, to form an air-tightened front air chamber of which a part except for an opening for sound output is spatially blocked from the outside on a front side on which the vibration plate of the driver unit is provided, and to form a rear air chamber that has a predetermined capacity on a rear side that is the opposite side to the front side; and an acoustic tube provided in a partial region of a partition wall of the housing that constitutes the rear air chamber and configured to spatially connect the rear air chamber and the outside of the housing through a tube.

Abstract: Disclosed is an audio signal processing method. The audio signal processing method according to the present invention comprises the steps of: receiving a bit-stream including at least one of a channel signal and an object signal; receiving a user's environment information; decoding at least one of the channel signal and the object signal on the basis of the received bit-stream; generating the user's reproducing channel information on the basis of the user's received environment information; and generating a reproducing signal through a flexible renderer on the basis of at least one of the channel signal and the object signal and the user's reproducing channel information.

Abstract: Some audio processing methods may involve receiving audio data corresponding to a plurality of audio channels and determining audio characteristics of the audio data, which may include transient information. An amount of decorrelation for the audio data may be based, at least in part, on the audio characteristics. If a definite transient event is determined, a decorrelation process may be temporarily halted or slowed. Determining transient information may involve evaluating the likelihood and/or the severity of a transient event. In some implementations, determining transient information may involve evaluating a temporal power variation in the audio data. Explicit transient information may or may not be received with the audio data, depending on the implementation. Explicit transient information may include a transient control value corresponding to a definite transient event, a definite non-transient event or an intermediate transient control value.

Abstract: To enable an output sound of a speaker unit to be output as a surface sound source in a limited space. A three-dimensional knitted fabric is provided at a position substantially facing speaker units while being given tension in a surface direction. Since the tension is applied, a pair of ground knitted fabrics constituting the three-dimensional knitted fabric and a connecting yarn extending back and forth between the ground knitted fabrics vibrate due to their elastic action. In particular, string vibration is generated in the connecting yarn. Due to this vibration, output sounds of the speaker units are propagated in the surface direction of the three-dimensional knitted fabric, and as a result, the three-dimensional knitted fabric propagates and radiates acoustic waves of a surface sound source.

Abstract: A processor can be associated with a memory for storing convolution data. A plurality of M filters from a corresponding plurality of M input channels to a selected one output channel can be provided, wherein each filter can be represented by a corresponding index, m. Each of the M filters can be partitioned into K respective filter partitions, wherein each respective filter partition can be represented by a corresponding index, k. A frequency-domain representation of each filter partition can be provided, wherein each frequency-domain representation of a filter partition comprises N frequency bins and a corresponding frequency-domain filter coefficient, wherein each respective frequency bin can be represented by a corresponding index, n. The memory can store such information in an arrangement suitable for the processor to concurrently receive sufficient information to concurrently convolve a frame of each input signal with the respective filters.

Abstract: Embodiments of the example embodiment relate to audio object extraction. A method for audio object extraction from audio content is disclosed. The method comprises determining a sub-band object probability for a sub-band of the audio signal in a frame of the audio content, the sub-band object probability indicating a probability of the sub-band of the audio signal containing an audio object. The method further comprises splitting the sub-band of the audio signal into an audio object portion and a residual audio portion based on the determined sub-band object probability. Corresponding system and computer program product are also disclosed.

Abstract: The present disclosure provides methods, devices and computer program products which provide less complex and more flexible control of the introduced decorrelation in an audio coding system. According to the disclosure, this is achieved by calculating and using two weighting factors, one for an approximated audio object and one for a decorrelated audio object, for introduction of decorrelation of audio objects in the audio coding system.

Abstract: In some embodiments, methods for generating an object based audio program including screen-elated metadata indicative of at least one warping degree parameter for at least one audio object, or generating a speaker channel-based program including by warping audio content of an object based audio program to a degree determined at least in part by at least one warping degree parameter, or methods for decoding or rendering any such audio program. Other aspects are systems configured to perform such audio signal generation, decoding, or rendering, and audio processing units (e.g., decoders or encoders) including a buffer memory which stores at least one segment of any such audio program.

Abstract: An audio echo suppressor includes a first receiver for receiving a first audio signal for rendering by a loud-speaker and a second receiver for receiving a microphone signal. A linear echo-cancellation filter generates a first compensation signal from the first audio signal and a compensator generates a residual signal by compensating the microphone signal for the first compensation signal. A first adapter determines filter parameters for the linear echo-cancellation filter. An estimator generates distortion measures where each distortion measure is indicative of the contribution to the residual signal in a frequency interval outside a first frequency interval. The residual signal results from rendering of signal components of the first audio signal that are within the first frequency interval. An echo reducer performs echo suppression based on the distortion measures.

Abstract: An audio processing system, such as an upmixer, may be capable of separating diffuse and non-diffuse portions of N input audio signals. The upmixer may be capable of detecting instances of transient audio signal conditions. During instances of transient audio signal conditions, the up-mixer may be capable of adding a signal-adaptive control to a diffuse signal expansion process in which M audio signals are output. The upmixer may vary the diffuse signal expansion process over time such that during instances of transient audio signal conditions the diffuse portions of audio signals may be distributed substantially only to output channels spatially close to the input channels. During instances of non-transient audio signal conditions, the diffuse portions of audio signals may be distributed in a substantially uniform manner.

Abstract: The present technology relates to an encoding device and method, a decoding device and method, and a program which can obtain high quality audio with a less code amount. A signal encoding unit encodes audio signals and outputs a resultant signal code string. A coefficient encoding unit encodes mixing coefficients for use in a mixing process of the audio signals and outputs a resultant coefficient code string. A multiplexing unit multiplexes the signal code string and the coefficient code string and outputs a resultant output code string. The coefficient encoding unit rearranges the mixing coefficients at the time of encoding the mixing coefficients on the basis of distances between input-side sound source positions and speaker positions on a reproduction side and calculates differential values between the mixing coefficients on the basis of arrangement order of the mixing coefficients, thereby encoding the mixing coefficients.

Abstract: A video conferencing system contains one or more display adjusting components, whereby an object to be displayed can be adjusted to appropriately fit various sized display screens. A display adjusting component is contained within the sending client, which adjusts the image of the object to be appropriately displayed to one or more receiving clients. The receiving clients also contain a display adjusting component, which can further adjust the image of the object to be displayed, as necessary. The multimedia conferencing server of the video conferencing system also contains a display adjusting component, which negotiates parameters of the sending and receiving clients. Any of the display adjusting components can function alone, or in any combination together. A method, and computer-readable media which contain computer-readable instructions to perform a method, of adjusting an image for video conference display are also described.