A method and system dynamically adjusts the audio of an audio and video signal to improve its overall sound quality and dialog intelligibility. Some embodiments use gain, equalization, audio signal compression and spatial enhancement (reverb) on individual channels of a multichannel audio signal.

An input audio signal is separated into input audio signal components (X(k,b)). A set of two or more band branches provides output audio signal components (Y(k,b)). The set of band branches comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal and the summed audio signal(y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

An audio artifact reduction technique operates on a pair of temporally-adjacent audio tracks. Samples at an end portion of a first track may be altered according to an inter-track sample discontinuity at a boundary between the two tracks. Samples at a beginning portion of a second track also may be altered according to the inter-track discontinuity. These altering operations may cause the sample values and the end of the first track and the beginning of the second track to be equal to each other at the boundary between the two tracks. In one aspect, the altering may generate adjustment curves that alter sample values of the tracks around the boundary between them. In another aspect, the altering may trim the tracks at zero crossings of track sample values nearest to the boundary between the tracks.

A system and method of generating binaural signals includes receiving, by a processing device, a sound signal including speech and noise components, and transforming, by the processing device using a deep neural network (DNN), the sound signal into a first signal and a second signal. The transforming further includes encoding, by an encoding layer of the DNN, the sound signal into a sound signal representation in a latent space, rendering, by a rendering layer of the DNN, the sound signal representation into a first signal representation and a second signal representation in the latent space, and decoding, by a decoding layer of the DNN, the first signal representation into the first signal and the second signal representation into the second signal.

A method and system dynamically adjusts the audio of an audio and video signal to improve its overall sound quality and dialog intelligibility. Some embodiments use gain, equalization, audio signal compression and spatial enhancement (reverb) on individual channels of a multichannel audio signal.

An enhanced Stereo Dimensional Array loudspeaker system 250 preferably including a mirror image pair of loudspeaker enclosures 280L, 280R configurable by a user or installer as a left-channel loudspeaker and a right channel loudspeaker each having a driver array aiming configuration with first and second angled baffle facets carrying main and effects drivers on separate facets and a Head Shadow filter signal processing system and method for driving the main and effects drivers to achieve a psycho-acoustically expanded image breadth by Head Shadow filter compensated inter-aural crosstalk cancellation.

A signal processing device includes: a resonant band-pass filter which extracts a frequency component which causes resonance of a housing from an input signal; a harmonic generating unit configured to generate a harmonic signal for an output signal of the resonant band-pass filter; an adding unit configured to add an output signal of the harmonic generating unit and at least a portion of a frequency component of the input signal; and a resonant band-stop filter which removes the frequency component which causes the resonance of the housing from an output signal of the adding unit.

An audio system for use in a confined space comprises N acoustic transducers, where N is an integer and N2. Each acoustic transducer is configured to receive an audio signal for a respective channel of the audio system and to generate an acoustic signal therefrom, and is operably coupled to a respective filter having a phase response. The phase response of at least one of the filters is different from the phase response of at least one other of the filters at a frequency f having a corresponding wavelength =d/(n+), where n is an integer and n0 and d is a characteristic separation of two or more of the N acoustic transducers when in the confined space. Compensation is thereby achieved for a reduction in acoustic power that would otherwise occur over a range of frequencies of width f around frequency f.

A device (160) and a method are proposed for tuning a frequency-dependent attenuation stage (122), with the purpose of suppressing non-linear distortion occurring in an audio reproduction system (120) associated with the frequency-dependent attenuation stage. The device comprises a receiving section (162) adapted to receive data representing an output acoustic signal (140) from the audio reproduction system, recorded upon excitation of the audio reproduction system by a predetermined input data signal (110)a first distortion detection section (163) adapted to detect presence of non-linear distortion based on the received data and to apply psycho-acoustic compensation to the detected non-linear distortion and a control section (164) adapted to determine, based on the psycho-acoustically compensated non-linear distortion, control information (170) suitable for controlling the frequency-dependent attenuation stage.

An input audio signal (101) is separated into input audio signal components (X(k,b)). A set of two or more band branches (105e) provides output audio signal components (Y(k,b)). The set of band branches (105e) comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal (101) and the summed audio signal (y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal (103) is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).