An apparatus and method for providing a musical instrument effects unit with transistor bias voltage visualization are disclosed. An example embodiment includes a musical instrument effects apparatus with transistor bias voltage visualization, the apparatus comprising: a transistor amplifier; a bias control coupled to the transistor amplifier to adjust an active bias voltage setting for the transistor amplifier; and a bias voltage meter coupled to the transistor amplifier to monitor, measure, and enable visualization of the active bias voltage setting, the bias voltage meter including a visualization device to produce a visual representation of the active bias voltage setting.

A neural network is trained to digitally model a reference audio system. Training is carried out by repeatedly performing a set of operations. The set of operations includes predicting by the neural network, a model output based upon an input, where the output approximates an expected output of the reference audio system, and the prediction is carried out in the time domain. The set of operations also includes applying a perceptual loss function to the neural network based upon a determined psychoacoustic property, wherein the perceptual loss function is applied in the frequency domain. Moreover, the set of operations includes adjusting the neural network responsive to the output of the perceptual loss function. A neural model file is output that can be loaded to generate a virtualization of the reference audio system.

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.

Disclosed are a method and device for processing an audio signal, in which a pitch processed signal 21 is mixed 33 with a high pass filtered 30 version of the input signal. This produces improvements in the latency and quality of the pitch processed signal, particularly for live performance.

A method and system for shaping the tone by modular circuit bending and modding of electric and electronic music instruments, audio amplifiers and sound equipment. The design of the system consists of a modding structure of a plurality of inserts, inputs and outputs connections for compatible modules, to reconfigure the electronic circuit without the physical changing of component values and without adding or removing electronic parts of the audio equipment by soldering. A plurality of tone modules for shaping the sound by the intervention into the signal path and a plurality of modules for shaping the tone by other sound design methods that are not part of the signal path are provided. For the modular system, an external device can be used as a multiplier, which consists of a plurality of inserts, inputs and outputs connections for compatible modules also including a plurality of combined series/parallel circuits of freely selectable inputs and outputs for the direct swap of a plurality of modding modules. In addition, switching options are available to control bypass, for advanced series or parallel connections, and for AB testing capabilities. The modular modding system can be used as a standalone device or as an upgrade to existing equipment.

A process is provided for training a neural network that digitally models an audio system. A sound source is utilized to electrically couple a test signal into an input of a reference audio system. The output of the reference audio system is collected into an audio interface coupled to a computer. A neural network is then trained using the test signal and the captured information to derive a set of weight vectors with appropriate values such that the overall output of the neural network converges towards an output representative of the reference audio system, and a signal in the time domain from a musical instrument is processed through the trained neural network with a latency under 20 milliseconds. A graphical user interface then outputs a graphical representation of the trained neural network, where the graphical representation visually displays at least one virtual control for interaction by a user.

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.

An amplifier and a method of outputting a waveform of a music signal capable of outputting a waveform of a music signal exceeding a power supply voltage is provided. An amplifier includes a power supply, an input terminal for a music signal, an amplifying circuit which amplifies the music signal using the power supply, and a jumping-up circuit which is connected to an output end of the amplifying circuit and outputs a waveform exceeding a voltage value of the power supply.

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.