Abstract: The present invention relates to audio signal processing, and more particularly to methods and apparatuses for emulating and controlling various features of mechanical spring reverberation in a digital audio processing system. According to certain aspects of the invention, such an emulation is performed so as to enhance or alter the characteristics of a digitally stored or processed audio signal in substantially the same manner as a mechanical spring reverberation system. In one example embodiment, the propagation of energy through a mechanical spring is simulated using dispersive waveguides, wherein left-going and right-going waves are separately processed, and the effects of dispersion and attenuation commuted to the waveguide ends. According to additional aspects, many spring reverberators contain spring elements arranged in parallel, with no coupling between springs.

Abstract: A system and method for graphic equalization of audio signals is disclosed. Traditional graphic equalizers provide control over the gains in each of a set of frequency bands. However, the actual band gains vary from the desired gains due to crosstalk between bands. Prior art methods for addressing this difficulty include applying a correction filter to the equalizer, and adjusting the shape of the individual band filters, both of which increase the computational cost. In an embodiment of the present invention, the input gains are processed to produce a set of adjusted gains which take into account the crosstalk, and result in an equalization interpolating the input band gains.

Abstract: A method and system for designing a discrete-time filter having a transfer function which approximates that of an analog shelf filter is disclosed. Prior art methods include applying the bilinear transform to the analog filter, which has the drawback of warping high-frequency features of the desired transfer function. In an embodiment of the present invention, an analog filter is designed which anticipates the warping imposed by the bilinear transform. For filters whose features approach the Nyquist limit, the inventive method provides a closer approximation to the analog response than direct application of the bilinear transform.

Abstract: A voice replacing method and system that uses the volume of the karaoke user's voice to control the volume of the lead singer's original vocal outputted to the audience. The volume of an original lead singer's voice is determined by the volume or vocal-amplitude level of the karaoke user's voice. The vocal-amplitude of the karaoke user is sensed using a vocal sensor. A vocal-amplitude digital waveform is created corresponding to the vocal-amplitude. A smoothed exponential waveform is created byan amplitudefollower corresponding to the digital waveform. The volume of the original lead singer's voice is adjusted by a sliding balance according to the smoothed exponential waveform. This exponential waveform allows for a continuous and smooth volume adjustment of the original lead singer's voice according to changes in volume of the karaoke user's voice.

Abstract: A system and method are disclosed for independently adjusting the pitch and timbre of an input signal within a modified Lent shifter. A method of adjusting the pitch and timbre of an input signal includes receiving an input signal and determining a window length corresponding to the input signal. A finite length spectrally smoothed signal is created from the input signal by synchronously windowing the input signal. The timbre of the finite length spectrally smoothed signal is adjusted thereby creating a finite length timbre adjusted signal. The finite length timbre adjusted signal is recombined at a rate necessary to produce a desired output pitch thereby reducing artifacts introduced by processing the finite length timbre adjusted signal and minimizing latency between receiving the input signal and producing a desired output signal.