An audio signal processing method, device and storage medium, are provided. The method includes performing sub-band filtering on a to-be-processed audio signal to obtain a plurality of sub-band signals, wherein the number of the sub-band signals is determined according to a lowest frequency of a band-pass filter and a cut-off frequency of an audio apparatus, and the sub-band signals comprise sub-band band-pass signals; and obtaining a target audio signal according to each of the sub-band band-pass signals and a processing algorithm of virtual bass enhancement signal.

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 pickup, a stringed instrument, and a pickup control method are disclosed. The pickup comprises: pickup components(110), multi-touch screens(130) and processing components(140). The multi-touch screens(130) and the processing components(140) are connected. The pickup components(110) and the processing components(140) are connected. The pickup components(110) is configured to pick up sound information emitted by a musical instrument. The multi-touch screen(130) is configured to display function application information that is selected from a group consisting of timbre adjustment function application information, tone adjustment function application information, and equalizer adjustment function application information. The multi-touch screen is configured to receive a function triggering instruction. The multi-touch screen displays information as to functional applications available through the pickup capabilities. Control functions available to a user are expanded beyond control by a single knob as in the prior art, the playing experience of a user on the instrument is enhanced.

A musical sound signal generation device continuously connects any one of a connected zeroth delay unit and a connected second delay unit to a fractional delay block and connects at least any one of a new zeroth delay unit and a new second delay unit to at least any one of the fractional delay block other than the fractional delay block connected to a new first delay unit in response to setting any one of the connected zeroth delay unit and the connected second delay unit as the new first delay unit, setting a delay unit in a preceding stage of the new first delay unit as the new zeroth delay unit, and setting a delay unit in a subsequent stage of the new first delay unit as the new second delay unit in accordance with a change in a designated tone pitch.

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.

Technologies are described for identifying familiar or interesting parts of music content by analyzing changes in vocal power using frequency spectrums. For example, a frequency spectrum can be generated from digitized audio. Using the frequency spectrum, the harmonic content and percussive content can be separated. The vocal content can then be separated from the harmonic and/or percussive content. The vocal content can then be processed to identify surge points in the digitized audio. In some implementations, the vocal content is included in the harmonic content during the separation procedure and is then separated from the harmonic content.

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.

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.

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.

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.