A specialized audio/instrument cable adapter with built-in digital signal processing capabilities that adds user-defined audio effects (such as reverb, delay, chorus and/or distortion) from within the cable adapter itself to affect the sound generated from an instrument or microphone such that the cable adapter (with a instrument/microphone cable) is the only connection needed between the instrument or microphone and an output device (such as an amplifier, PA, powered speaker, music mixer, or a recording device). The audio effects used by the cable adapter can be changed via (i) an app from a smartphone, tablet, computer or other electronic device; (ii) a wireless controller that attaches to the instrument; (iii) a pedal, and/or (iv) any other type of wireless controller that has the ability to communicate with a smartphone/tablet/computer or other electronic device.

Vocal musical performances may be captured and, in some cases or embodiments, pitch-corrected and/or processed in accord with a user selectable vocal effects schedule for mixing and rendering with backing tracks in ways that create compelling user experiences. In some cases, the vocal performances of individual users are captured on mobile devices in the context of a karaoke-style presentation of lyrics in correspondence with audible renderings of a backing track. Such performances can be pitch-corrected in real-time at the mobile device in accord with pitch correction settings. Vocal effects schedules may also be selectively applied to such performances. In these ways, even amateur user/performers with imperfect pitch are encouraged to take a shot at “stardom” and/or take part in a game play, social network or vocal achievement application architecture that facilitates musical collaboration on a global scale and/or, in some cases or embodiments, to initiate revenue generating in-application transactions.

A specialized audio/instrument cable with built-in digital signal processing capabilities that adds user-defined audio effects (such as reverb, delay, chorus and/or distortion) from within the cable itself to affect the sound generated from an instrument or microphone such that the cable is the only connection needed between the instrument or microphone and an output device (such as an amplifier, PA, powered speaker, music mixer, or a recording device). The audio effects used by the cable can be changed via (i) an app from a smartphone, tablet, computer or other electronic device; (ii) a wireless controller that attaches to the instrument; (iii) a pedal, and/or (iv) any other type of wireless controller that has the ability to communicate with a smartphone/tablet/computer or other electronic device.

Preparing a sum of complex exponentials in the form of a modal filter to approximate a given impulse response is considered, and a non-iterative solution is described. In one embodiment, the mode count, and the mode frequencies, dampings, and amplitudes are estimated using the generalized eigenvalues of Hankel matrices of samples of the given impulse response.

A system for enhancing artificial reverberation in a noisy listening space includes a sensor configured to generate a signal indicating a current noise level in a listening space, a loudspeaker configured to output sound in the listening space based on an output signal, and at least one processor. The at least one processor is configured to execute instructions to: generate an artificial reverberation signal based on a source signal and a response of a listening space, wherein the artificial reverberation signal includes a first low output-level portion having output levels that are below a compression threshold level and a first high output-level portion having output levels that are above the compression threshold level; generate a compressed artificial reverberation signal based on the artificial reverberation signal by increasing the output levels of the low output-level portion by a first magnitude and increasing the output levels of the high output-level portion by less than the first magnitude; and combine the compressed artificial reverberation signal with the source signal to form an output signal for the listening space.

There is provided a signal processing apparatus that includes a control unit that executes, on a basis of a waveform signal generated in accordance with a motion of an attachment portion of a sensor attached to a tool or a body, effect processing for the waveform signal or another waveform signal, the waveform signal being output from the sensor.

Systems and methods for generating a haptic output from an audio signal having a continuous stream of sampled digital audio data are provided. A haptic processing system receives the digital audio data, analyses the digital audio data for processing and extracts haptic signals for generating a haptic effect through an actuator. The method includes passing the digital audio signal on through dynamic processor(s), adjusting the dynamic range of the digital audio signal, extracting the signal envelope of the audio data, synthesising low-frequency signals from the extracted signal envelope, and enhancing the low-frequency content using a resonator. The haptic output is generated by mixing the digital audio signal with outputs from the different modules of the haptic processing system. An analytics module monitors, controls and adjusts the processing of the digital audio signal at the noise gate module, the compressor module and the envelope module to enhance the haptic output.

Methods and systems for performing modal reverb techniques for audio signals are described. The method may involve simplifying a reverb effect to be applied to the audio signal by receiving an IR, dividing the IR into a plurality of sub-bands, using a parametric estimation algorithm to determine respective parameters of the modes included in each sub-band, aggregating the respective modes of the sub-bands into a set; and truncating the set of aggregated modes into a subset of modes. Reverberation of the audio signal may be manipulated based on an IR that itself is based on the truncated subset of modes.

The disclosed computer-implemented method may include obtaining an audio sample from a content source, inputting the obtained audio sample into a trained machine learning model, obtaining the output of the trained machine learning model, wherein the output is a profile of an environment in which the input audio sample was recorded, obtaining an acoustic impulse response corresponding to the profile of the environment in which the input audio sample was recorded, obtaining a second audio sample, processing the obtained acoustic impulse response with the second audio sample, and inserting a result of processing the obtained acoustic impulse response and the second audio sample into an audio track. Various other methods, systems, and computer-readable media are also disclosed.

The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.