A specialized audio/instrument cable with built-in digital signal processing capabilities that adds digital audio sampling capabilities that allows the user to trigger synthesized sounds or virtual musical instruments 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. Using voice recognition, the specialized cable can select an audio effects chain algorithm and/or sampled sound algorithm extrapolated from a musical digital audio fingerprint (MDAF) created from a desired musician's instrument to alter the sound of the input instrument's audio.

The present disclosure provides a method for chorus mixing, an apparatus, an electronic device and storage media. The method includes converting a main vocal audio signal and a chorus audio signal into signals in frequency domain, respectively, wherein the chorus audio signal comprises main vocal audio played by a speaker; determining a delay between the main vocal audio signal and the chorus audio signal based on a frequency-domain signal of the main vocal audio signal and a frequency-domain signal of the main vocal audio played by the speaker included in a frequency-domain signal of the chorus audio signal; aligning the chorus audio signal with the main vocal audio signal based on the determined delay; performing an echo cancellation on the aligned chorus audio signal; and mixing audio of the main vocal audio signal and the echo-canceled chorus audio signal.

The present embodiments relate to audio effect processing, and more particularly to a method for creating reverberation impulse responses from prerecorded or live source materials forms the basis of a family of reverberation effects. In one embodiment, segments of audio are selected and processed to form an evolving sequence of reverberation impulse responses that are applied to the original source material—that is, an audio stream reverberating itself. In another embodiment, impulse responses derived from one audio track are applied to another audio track. In a further embodiment, reverberation impulse responses are formed by summing randomly selected segments of the source audio, and imposing reverberation characteristics, including reverberation time, wet equalization, wet-dry mix, and predelay. By controlling the number and timing of the selected source audio segments, the method produces a collection of impulse responses that represent a trajectory through the source material. In so doing, the evolving impulse responses will have the character of room reverberation while also expressing the changing timbre and dynamics of the source audio.

An electronic musical instrument according to one embodiment includes: a sound source configured to generate a first sound signal and a second sound signal in accordance with an instruction signal for instructing to produce a sound; a first output unit configured to output a third sound signal containing the first sound signal and the second sound signal at a first sound volume ratio; and a second output unit configured to output a fourth sound signal containing the first sound signal and the second sound signal at a second sound volume ratio that is different from the first sound volume ratio.

An electronic device having circuitry configured to perform source separation on an audio signal to obtain a separated source and a residual signal, to perform feature extraction on the separated source to obtain one or more processing parameters, and to perform audio processing on a captured audio signal based on the one or more processing parameters to obtain an adjusted separated source.

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.

The present invention discloses an integrated karaoke device, including a microphone and a sound box. The sound box includes a sound chamber containing a loudspeaker. The integrated karaoke device further includes a connecting part made of flexible material that is fixedly connected between a bottom of the microphone and a top of the sound chamber. The use of a flexible material connecting part rather than a solid structural part reduces the transmission to the microphone of sound vibration generated by the sound box. The beneficial effect of implementing the present invention is that a flexible material connecting part is fixedly connected between the bottom of the microphone and the top of the sound chamber, and meanwhile, a gap is kept between the bottom of a printed circuit board (PCB) and the top of the sound chamber; and because the flexible material connecting part can effectively reduce sound vibration, the transmission of vibration to the microphone from the sound box can be effectively eliminated, thereby preventing squealing, and enabling integration of the microphone and the sound box for use in karaoke.

A small (smartphone-sized or tablet-sized), foot-enabled, flat, tiltable, rotatable, dynamic touch screen pedal and controller that uses a tilt mechanism to toggle between and select different audio (or other) functions and effects that are displayed on the attached display. The tilt of the device as well as optional tapping sequences activate different functions in predetermined function locations. Specific audio effects, audio effect presets, loops, songs, and controller functions can each be assigned to different touch screen locations, pedal buttons, and/or tilt directions.

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.

Provided is a method for processing audio including: acquiring an accompaniment audio signal and a voice signal of a current to-be-processed musical composition; determining a target reverberation intensity parameter value of the acquired accompaniment audio signal, wherein the target reverberation intensity parameter value is configured to indicate a rhythm speed, an accompaniment type, and a performance score of a singer of the current to-be-processed musical composition; and reverberating the acquired vocal signal based on the target reverberation intensity parameter value.