Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal; (b) a bass extraction filter that extracts a bass portion of such original audio signal; (c) an estimator that estimates a fundamental frequency of a bass sound within such bass portion; (d) a frequency translator that shifts the bass portion by a positive frequency increment that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (f) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (g) an audio output device coupled to the output of the adder.

Multiple example valve amplifiers are provided. A first example valve amplifier is provided which comprises (i) a valve power amplifier switchable between a high-power mode and a low-power mode and (ii) a loudspeaker simulator circuit, the valve amplifier being configured such that the valve power amplifier drives the loudspeaker simulator circuit in the low-power mode. A second example valve amplifier is provided which comprises a switched-mode power supply, SMPS, system), the SMPS system comprising (i) an SMPS and (ii) circuitry configured to enable an output impedance of the SMPS to be switched between first and second output impedances, the first output impedance being lower than the second output impedance.

Systems and methods for providing accurate and independent control of reverberation properties are disclosed. In some embodiments, a system may include a reverberation processing system, a direct processing system, and a combiner. The reverberation processing system can include a reverb initial power (RIP) control system and a reverberator. The RIP control system can include a reverb initial gain (RIG) and a RIP corrector. The RIG can be configured to apply a RIG value to the input signal, and the RIP corrector can be configured to apply a RIP correction factor to the signal from the RIG. The reverberator can be configured to apply reverberation effects to the signal from the RIP control system. In some embodiments, one or more values and/or correction factors can be calculated and applied such that the signal output from a component in the reverberation processing system is normalized to a predetermined value (e.g., unity (1.0)).

A computer-implemented method of processing audio data, the method comprising receiving input audio data (x) comprising a time-series of amplitude values; transforming the input audio data (x) into an input frequency band decomposition (X1) of the input audio data (x); transforming the input frequency band decomposition (X1) into a first latent representation (Z); processing the first latent representation (Z) by a first deep neural network to obtain a second latent representation (Z{circumflex over ( )}, Z1{circumflex over ( )}); transforming the second latent representation (Z{circumflex over ( )}, Z1{circumflex over ( )}) to obtain a discrete approximation (X3{circumflex over ( )}); element-wise multiplying the discrete approximation (X3{circumflex over ( )}) and a residual feature map (R, X5{circumflex over ( )}) to obtain a modified feature map, wherein the residual feature map (R, X5{circumflex over ( )}) is derived from the input frequency band decomposition (X1); processing a pre-shaped frequency band decomposition by a waveshaping unit to obtain a waveshaped frequency band decomposition (X1{circumflex over ( )}, X1.2{circumflex over ( )}), wherein the pre-shaped frequency band decomposition is derived from the input frequency band decomposition (X1), wherein the waveshaping unit comprises a second deep neural network; summing the waveshaped frequency band decomposition (X1{circumflex over ( )}, X1.2{circumflex over ( )}) and a modified frequency band decomposition (X2{circumflex over ( )}, X1.1{circumflex over ( )}) to obtain a summation output (X0{circumflex over ( )}), wherein the modified frequency band decomposition (X2{circumflex over ( )}, X1.1{circumflex over ( )}) is derived from the modified feature map; and transforming the summation output (X0{circumflex over ( )}) to obtain target audio data (y{circumflex over ( )}).

A computer-implemented method of processing audio data, the method comprising receiving input audio data (x) comprising a time-series of amplitude values; transforming the input audio data (x) into an input frequency band decomposition (X1) of the input audio data (x); transforming the input frequency band decomposition (X1) into a first latent representation (Z); processing the first latent representation (Z) by a first deep neural network to obtain a second latent representation (Z{circumflex over ( )}, Z1{circumflex over ( )}); transforming the second latent representation (Z{circumflex over ( )}, Z1{circumflex over ( )}) to obtain a discrete approximation (X3{circumflex over ( )}); element-wise multiplying the discrete approximation (X3{circumflex over ( )}) and a residual feature map (R, X5{circumflex over ( )}) to obtain a modified feature map, wherein the residual feature map (R, X5{circumflex over ( )}) is derived from the input frequency band decomposition (X1); processing a pre-shaped frequency band decomposition by a waveshaping unit to obtain a waveshaped frequency band decomposition (X1{circumflex over ( )}, X1.2{circumflex over ( )}), wherein the pre-shaped frequency band decomposition is derived from the input frequency band decomposition (X1), wherein the waveshaping unit comprises a second deep neural network; summing the waveshaped frequency band decomposition (X1{circumflex over ( )}, X1.2{circumflex over ( )}) and a modified frequency band decomposition (X2{circumflex over ( )}, X1.1{circumflex over ( )}) to obtain a summation output (X0{circumflex over ( )}), wherein the modified frequency band decomposition (X2{circumflex over ( )}, X1.1{circumflex over ( )}) is derived from the modified feature map; and transforming the summation output (X0{circumflex over ( )}) to obtain target audio data (y{circumflex over ( )}).

A system generates a haptic signal. The system receives an audio signal, pre-processes the audio signal by modifying the audio signal to create a modified audio signal, and maps the modified audio signal to a haptic signal. The system then sends the haptic signal to an actuator to generate one or more haptic effects.

Systems and methods for providing accurate and independent control of reverberation properties are disclosed. In some embodiments, a system may include a reverberation processing system, a direct processing system, and a combiner. The reverberation processing system can include a reverb initial power (RIP) control system and a reverberator. The RIP control system can include a reverb initial gain (RIG) and a RIP corrector. The RIG can be configured to apply a RIG value to the input signal, and the RIP corrector can be configured to apply a RIP correction factor to the signal from the RIG. The reverberator can be configured to apply reverberation effects to the signal from the RIP control system. In some embodiments, one or more values and/or correction factors can be calculated and applied such that the signal output from a component in the reverberation processing system is normalized to a predetermined value (e.g., unity (1.0)).

Provided is a vibration system using sound. More particularly, the present invention relates to a system which is capable of generating vibration using sound so that the beats of the sound may be felt, is convenient to carry or move due to a lightweight and compact size thereof, is capable of generating vibration matching a beat of sound to which a user is currently listening, is capable of generating vibrations matching various feelings according to beats of sound on the basis of user setting, thereby greatly enhancing effects that the user may feel, and is very inexpensive to manufacture.

A method of outputting a parameter of a sound processing device receives an audio signal, obtains information of the parameter of the sound processing device, which corresponds to the received audio signal, by using a trained model obtained by performing training of a relationship among a training output sound of the sound processing device, a training input sound of the sound processing device, and a parameter of sound processing performed by the sound processing device, the parameter of the sound processing device being receivable by a user of the sound processing device, and outputs obtained information of the parameter of the sound processing device corresponding to the received audio signal.

A method of outputting a parameter of a sound processing device receives an audio signal, obtains information of the parameter of the sound processing device, which corresponds to the received audio signal, by using a trained model obtained by performing training of a relationship among a training output sound of the sound processing device, a training input sound of the sound processing device, and a parameter of sound processing performed by the sound processing device, the parameter of the sound processing device being receivable by a user of the sound processing device, and outputs obtained information of the parameter of the sound processing device corresponding to the received audio signal.