A method and apparatus for audio signal processing in an audio chain to correct a non-linearity of the electroacoustic transducers in the audio chain by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain using an approximation of the quadratic function. The method accommodates the psychoacoustical characteristics of the human ear by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain approximating by a non-linear fifth degree polynomial function for a pressure change by the human ear up to p_Δ. The method and apparatus reduce non-linearities of the entire audio chain with the human ear, by adding non-linearities in the audio chain so that an audio chain characteristic reduces the non-linearity of the human ear polynomial approximation to the pressure change p_Δ=±1 Pa.

A dual microphone signal processing arrangement for reducing reverberation is described. Time domain microphone signals are developed from a pair of sensing microphones. These are converted to the time-frequency domain to produce complex value spectra signals. A binary gain function applies frequency-specific energy ratios between the spectra signals to produce transformed spectra signals. A sigmoid gain function based on an inter-microphone coherence value between the transformed spectra signals is applied to the transformed spectra signals to produce coherence adapted spectra signals. And an inverse time-frequency transformation is applied to the coherence adjusted spectra signals to produce time-domain reverberation-compensated microphone signals with reduced reverberation components.

According to an example embodiment, a technique for processing an input audio signal (101) comprising a multi-channel audio signal is provided, the technique comprising: deriving (104), based on the input audio signal (101), a first signal component (105-1) comprising a multi-channel audio signal that represents a focus portion of a spatial audio image conveyed by the input audio signal and a second signal component (105-2) comprising a multi-channel audio signal that represents a non-focus portion of the spatial audio image; processing (112) the second signal component (105-2) into a modified second signal component (113) wherein the width of the spatial audio image is extended from that of the second signal component (105-2); and combining (114) the first signal component (105-1) and the modified second signal component (112) into an output audio signal (115) comprising a multi-channel audio signal that represents partially extended spatial audio image.

A recurrent neural network is employed in a loudspeaker system to compensate the distortion of the system based upon a source signal (content) and the sensing output of a sensing circuit (context). A frequency domain transform is selected to provide mapping between the source signal and a recorded signal; and enable reconstruction of desirable playback. Various sensing-related features and source-related features are derived to serve as the auxiliary information. A desirable content is therefore generated based upon the original content and the context.

A phase optimizer optimizes, for each listening position in a listening environment, a phase shift for each frequency in a range of predetermined frequencies. The phase optimizer determines a resultant phase value for each possible phase shift value and stores the resultant phase values for each possible phase shift value in an array. The phase optimizer calculates mean and standard deviation for the resultant phases stored in the array. The mean and standard deviations stored in the array are compared and phase shift values that result in the resultant phase values having the smallest mean and standard deviations are selected and are stored in memory. The phase optimizer optimizes each frequency, within a predetermined range of frequencies, for all possible phase shift values within a predetermined range of phase shift values and generates a phase shift target curve generated to be output by the phase optimizer.

Disclosed is a method of presenting audio information to a user. The method comprising receiving samples of a waveform from a media handling component, initializing a biquad filter with a set of one or more coefficients corresponding to a set of one or more stages of the biquad filter for both a real component of the samples and an imaginary component of the samples. The biquad filter is implemented on a media processing component of the mobile media device. The method further comprises applying the biquad filter to the samples of the waveform to generate an output for presentation to the user, the output comprising a processed rendering of the real component of the samples and the imaginary component of the samples.

A method for generating sound within a predetermined environment, the method comprising: emitting a first audio signal from a first location; and concurrently emitting a second audio signal from a second location, wherein: the first location and second location are distinct within the environment; the first audio signal and second audio signal have the same frequency; and the first audio signal and second audio signal have a phase difference that varies as a function of time to limit the time-averaged interference fluctuation across the environment.

An apparatus for providing a contrast mode and a front optimized mode for audio in a vehicle is provided. An audio controller is programmed to transmit first audio content in a first zone seating area and to transmit second audio content in a second zone seating area. The audio controller receives a first indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the contrast mode to provide an equal listening experience. The audio controller receives a second indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the front optimized mode to increase a quality of sound in the first zone seating area and to decrease a quality of sound in the second zone seating area.

According to an example embodiment, a method for processing an input audio signal (101) in accordance with spatial metadata (103) so as to play back a spatial audio signal in a device (50) in dependence of at least one sound reproduction characteristic (105) of the device is provided, the method comprising obtaining said input audio signal (101) and said spatial metadata (103); obtaining said at least one sound reproduction characteristic (105) of the device; rendering a first portion of the spatial audio signal using a first type playback procedure applied on the input audio signal in dependence of the spatial metadata (103), wherein the first portion comprises sound directions within a front region of the spatial audio signal; and rendering a second portion of the spatial audio signal using a second type playback procedure applied on the input audio signal in dependence of the spatial metadata (103) and in dependence of said at least one sound reproduction characteristic (105), wherein the second portion comprises sound directions that are not included in the first portion and where the second type playback procedure is different from the first playback procedure and involves cross-talk cancellation processing.

The present technology relates to a signal processing device, a signal processing method, and a program for enabling reproduction of high-quality sounds with a low process load. The signal processing device includes a demultiplexing section that extracts encoded audio signals and overamplitude flags, which have been generated for a plurality of respective panel loudspeakers and each indicate whether overamplitude will occur in the corresponding panel loudspeaker, by demultiplexing encoded data, a decoding section that decodes the encoded audio signals, and an adjustment section that adjusts audio signals to be supplied to the plurality of panel loudspeakers on the basis of the overamplitude flags and audio signals resulting from the decoding. The present technology is applicable to an encoding device and a decoding device.