The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of Δf. The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of FΔf; with F being a resolution factor, with F≥1; wherein the transposition order P is different from the resolution factor F.

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 ( )}).

Disclosed are systems and methods for processing an audio signal. In particular, there is provided a method for determining dynamic gain values to be applied on a digital input signal. The digital signal may be arranged in blocks. The dynamic gain values may be used for attenuating input signal values exceeding a clipping threshold. More particularly, the method comprising, for each signal block, passing backwards over the next signal block and the current signal block to produce a preliminary gain contour from the input signal; and passing forwards over the current signal block to produce a final gain contour for the current signal block based on the preliminary gain contour, wherein the gain contours are produced by applying an instant gain ascent and a smooth gain decay to the gain contours.

A method in accord with certain implementations involves, at a data interface of a musical instrument effects processor, receiving an extracted characteristic of an audible sound that is captured at a microphone; transferring the extracted characteristic to a digital signal processor residing in the musical instrument effects processor; receiving input signals at an input to the musical instrument effects processor; at the digital signal processor of the musical instrument effects processor, modifying the received input signals using the extracted characteristics to create the electronic audio effect; and outputting the modified input signals as an output signal from the musical instrument effects processor. This abstract is not to be considered limiting, since other embodiments may deviate from the features described in this abstract.

A corrector is configured to transform audio signals to compensate for unwanted distortion characteristics of a loudspeaker. A tuning filter is configured to transform audio signals to incorporate desired distortion characteristics associated with a target loudspeaker. By chaining together the tuning filter and the corrector, an audio signal can be modified so that the loudspeaker, when outputting the audio signal, has response characteristics of the target loudspeaker.

Disclosed are systems and methods for processing an audio signal. In particular, there is provided a method for determining dynamic gain values to be applied on a digital input signal. The digital signal may be arranged in blocks. The dynamic gain values may be used for attenuating input signal values exceeding a clipping threshold. More particularly, the method comprising, for each signal block, passing backwards over the next signal block and the current signal block to produce a preliminary gain contour from the input signal; and passing forwards over the current signal block to produce a final gain contour for the current signal block based on the preliminary gain contour, wherein the gain contours are produced by applying an instant gain ascent and a smooth gain decay to the gain contours.

An audio device for amplifying an audio signal enables a user to instantly switch between two or more active amplifying components, thereby allowing the user to evaluate the effect of a particular active amplifying component in the audio device. In one example, the audio device is an audio vacuum tube preamplifier, and has a circuit configuration that facilitates A/B comparison between two or more preamp tubes by hot switching between preamp tubes in the audio signal path. The input to the two or more tubes is switched between each tube by the listener on demand without pause or interruption to the music or program.

The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of Δf. The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of FΔf; with F being a resolution factor, with F≥1; wherein the transposition order P is different from the resolution factor F.

A wireless controller that allows a musician to wirelessly switch between audio effect presets, loops, and/or songs which are stored in a specialized audio/instrument cable or adapter. The wireless controller can be located in the same place as (or incorporated into) a digital guitar tuner on a guitar headstock allowing the musician to operate the wireless controller in a familiar location while switching between audio effect presets, loops, and/or songs by tapping on small buttons.

The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of Δf. The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of FΔf; with F being a resolution factor, with F≥1; wherein the transposition order P is different from the resolution factor F.