G10H1/12

DEEP ENCODER FOR PERFORMING AUDIO PROCESSING

Embodiments are disclosed for determining an answer to a query associated with a graphical representation of data. In particular, in one or more embodiments, the disclosed systems and methods comprise receiving an input including an unprocessed audio sequence and a request to perform an audio signal processing effect on the unprocessed audio sequence. The one or more embodiments further include analyzing, by a deep encoder, the unprocessed audio sequence to determine parameters for processing the unprocessed audio sequence. The one or more embodiments further include sending the unprocessed audio sequence and the parameters to one or more audio signal processing effects plugins to perform the requested audio signal processing effect using the parameters and outputting a processed audio sequence after processing of the unprocessed audio sequence using the parameters of the one or more audio signal processing effects plugins.

Efficient combined harmonic transposition

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.

SYSTEMS AND METHODS FOR PROVIDING AUDIO-FILE LOOP-PLAYBACK FUNCTIONALITY

Systems and methods for providing audio-file loop-playback functionality are provided. The system includes a processor that performs a method including setting a playback loop start-point based on a first selection of a button; setting a loop end-point, associating a loop with an audio file, and entering into the loop based on a second selection of the button; and exiting the loop based on a third selection of the button. Associating the loop with the audio file includes adding metadata to the audio file. The metadata associates the loop with a button. The method includes reentering the loop based on a fourth selection of the button and exiting the loop based on a fifth selection of the button.

SYSTEMS AND METHODS FOR PROVIDING AUDIO-FILE LOOP-PLAYBACK FUNCTIONALITY

Systems and methods for providing audio-file loop-playback functionality are provided. The system includes a processor that performs a method including setting a playback loop start-point based on a first selection of a button; setting a loop end-point, associating a loop with an audio file, and entering into the loop based on a second selection of the button; and exiting the loop based on a third selection of the button. Associating the loop with the audio file includes adding metadata to the audio file. The metadata associates the loop with a button. The method includes reentering the loop based on a fourth selection of the button and exiting the loop based on a fifth selection of the button.

METHOD AND INSTALLATION FOR PROCESSING A SEQUENCE OF SIGNALS FOR POLYPHONIC NOTE RECOGNITION
20170365244 · 2017-12-21 · ·

This is a method and installation in which a time-domain digital audio signal is split into a plurality of narrow-band time-domain digital audio signals confined to specific frequency bands, short-term segments of which are temporarily stored in memory. The method comprises the use of signal processing algorithms for extracting multiple signal features from said short-term segments in a fixed sequence or upon request from a decision-making algorithm. Said decision-making algorithm makes tentative or final decisions about the type of occupancy of frequency bands resulting from the extracted features. Said decision-making algorithm may request from said signal processing algorithms further specific feature extractions from specific short-term segments and make further tentative or final decisions about the type of occupancy of frequency bands resulting from the requested features. Next, said decision-making algorithm stores its tentative decisions and makes final decisions about band occupancy for processing together with results from later short-term segments. Eventually, said decision-making algorithm outputs final decisions derived from current and past short-segments in the form of a set of notes having been played over some recent time interval, together with information as to the timing of each note from the set.

TIME-VARYING AND NONLINEAR AUDIO PROCESSING USING DEEP NEURAL NETWORKS

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 ( )}).

Apparatuses and methods for audio classifying and processing

Apparatus and methods for audio classifying and processing are disclosed. In one embodiment, an audio processing apparatus includes an audio classifier for classifying an audio signal into at least one audio type in real time; an audio improving device for improving experience of audience; and an adjusting unit for adjusting at least one parameter of the audio improving device in a continuous manner based on the confidence value of the at least one audio type.

SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD
20170353169 · 2017-12-07 ·

A signal processing apparatus includes: an obtaining portion configured to obtain an impulse response in a semi-open state in which, among a plurality of acoustic feedback systems, at least one acoustic feedback system is open and at least one acoustic feedback system is closed; and a calculating portion configured to calculate a gain correction amount based on the impulse response that the obtaining portion has obtained and to output a calculated gain correction amount.

DYNAMIC AND MODIFIABLE ENVIRONMENTAL SYSTEM
20230187022 · 2023-06-15 ·

A computing system that has an application that captures input and develops an integrated environment using biomes with user preferences that are replicated and transferred between different biomes. At least one processor of the computing device is capable of receiving contextual data related to an event and can determine at least one biome for analyzing the received context data and compares a first genetic strand to a second genetic strand related to data associated with the received context data. The processer also determines a third genetic strand, based on the first genetic strand and the second genetic strand, that is a new genetic strand of data different from the first and second genetic strands. The processor also s determines relevancy of the first and second strand and other biomes, and updates the computing system associated with the determined context.

Systems and methods for selecting an audio track by performing a gesture on a track-list image

Systems and methods for selecting an audio track by performing a gesture on a track-list image are provided. The system includes a processor that performs a method including displaying the audio-track list, detecting a contact with the touchscreen display at a location corresponding to the audio track, detecting a continuous movement of the contact in a direction, detecting a length of the continuous movement, and selecting the audio track if the continuous movement has a length longer than a threshold length. The method includes shifting text associated with the audio track based on the length and direction of the continuous movement. The method includes determining that the selection is a command to queue the audio track for playback or add it to a preparation track list. This determination may be based on the direction of the continuous movement.