The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker. The values obtained by multiplying these gains by the gain of the virtual speaker are set as the gains of the lower right and lower left speakers for fixing a sound image at the target sound image position. The present technology can be applied to sound processing apparatuses.

A frequency estimation apparatus and method for a single receiver. The frequency estimation apparatus for the single receiver includes: a coarse frequency estimation unit estimating a coarse frequency by calculating an average of frequency estimation values for each single pulse; a direct current domain transformation unit transforming a reception signal into a direct current domain based on the coarse frequency; a fine frequency estimation unit estimating a fine frequency by applying regression analysis to a pulse train in the direct current domain of the coarse frequency; and an extremely fine frequency estimation unit estimating an extremely fine frequency by compensating an error of the coarse frequency with the fine frequency.

An audio system includes a loudspeaker cabinet defining a longitudinal axis. Several loudspeaker transducers are distributed around the longitudinal axis. The audio system includes an audio rendering processor to cause the loudspeaker transducers to emit a sound field approximating a desired contour. The desired contour can be decomposed into a combination of several constituent modal beam components, and the audio rendering processor can render a truncated version of the decomposition to render an approximation of the desired contour. The desired contour can be one of a plurality of contours stored in a memory, or can be user defined. The cabinet includes a processor and a memory having instructions that, when executed by the processor, cause the audio system decompose a desired contour and to render a truncated version of the decomposition. Related principles are described by way of reference to method and apparatus examples.

An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

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.

Systems and methods for holistically modelling audio feedback and removing the entire feedback signal corresponding thereto. The systems can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume), than those conventional systems which seek to remove singing frequencies with PEQs. The systems are an improvement over traditional audio feedback elimination systems which attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling, and those feedback elimination systems which simply employ adaptive notch filtering to detect and notch the so-called singing or howling frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. The systems set forth herein obviate the need for tuning knobs or buttons, making set up easy.

A method to adjust audio amplification may include presenting an audio configuration interface. The audio configuration interface may include multiple amplification settings that each correspond to a different one of multiple frequency range. The multiple amplification settings indicating amplifications may be applied to the multiple frequencies of audio output by a device. The method may further include obtaining an action to adjust the amplification of one or more of multiple amplification settings in the audio configuration interface to generate adjusted amplification settings. The method also includes, in response to obtaining the action, automatically applying the adjusted amplification settings to test audio and automatically outputting the test audio with the applied adjusted amplification settings through a speaker of the device.

Disclosed herein are system, method, and tangible computer readable medium for creating a desired audio effect for a user. The method includes operations including: causing a plurality of speakers to play test signals, each test signal being specific to one of the speakers; receiving from a remote device recorded frequency responses of the speakers resulting from the playing of the test signals; creating one or more filters to match an audio profile selected by a user; applying the filters to the recorded frequency responses to obtain filtered transformations of the speakers; and transmitting the filtered transformations to the speakers; wherein the filtered transformations are applied at the speakers to thereby achieve the user audio profile.

The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image.

A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker. The values obtained by multiplying these gains by the gain of the virtual speaker are set as the gains of the lower right and lower left speakers for fixing a sound image at the target sound image position. The present technology can be applied to sound processing apparatuses.

Embodiments described herein provide for smart configuration of audio settings for a playback device. According to an embodiment, while a playback device is a part of a first zone group that includes the playback device and at least one first playback device, the playback device applies a first audio setting. The embodiment also includes the playback device joining a second zone group that includes the playback device and at least one second playback device. The embodiment further includes the playback device applying a second audio setting based on an audio content profile corresponding to the second zone group.