Chromatic Research, Inc.

Leland Stanford Junior University

Avery L Wang

Brooks S Read

A method for resampling includes convolving a given set of samples with the impulse response function of a low-pass filter. In this method, values of the impulse response required for the convolution calculation are computed at the time of resampling from a segmented polynomial approximating the impulse response.

Arbitrary-ratio sampling rate converter using approximation by segmented

A method and apparatus for producing a tone (e.g. for music) without use of a waveform memory and using a feedback loop. The feedback loop includes a waveform generator which calculates, in real time, a parabolic approximation to a sine wave. The feedback loop includes a delay phase differencer to eliminate hunting. The output waveform from the feedback loop is provided to a sine function generator which approximates a sine value using a third order polynomial, to provide the output tone.

Non-linear tone generator

A system and method for stack-based processing of multiple real-time tasks operates on a net list of tasks which operate essentially simultaneously with system resources shared between tasks in a dynamic configuration. This system and method operate to control dispatching of messages which activate signal processing tasks, sequencing of processes activated by the messages and management of signal flow. Tasks are dynamically activated and deactivated in accordance with the specification by the net list by manipulating the task signals on the stack, thereby substantially reducing high-speed memory requirements of the system.

System and method for stack-based processing of multiple real-time audio

A method for resampling includes convolving a given set of samples with the impulse response function of a low-pass filter. In this method, values of the impulse response required for the convolution calculation are computed at the time of resampling from a segmented polynomial approximating the impulse response. In one embodiment, the method is applied to provide musical tones of various pitches from a stored waveform.

Method for varying the pitch of a musical tone produced through playback

A quasi periodic signal is frequency warped by selectively frequency modulating it, thereby resulting in a signal that is stationary with a simplified spectrum which is more amenable to analysis. The resultant demodulated signal is low pass filtered resulting in an analytic signal whose phase winding rate is the frequency of the signal. The phase is differenced by multiplying the signal with a delayed version of itself creating an instantaneous autocorrelation. Thereafter the phase difference is measured with a complex arctangent to yield a resulting phase error. The resulting phase error is integrated by an integrator whose output value is the estimate of the frequency. This output frequency parameter is then used to update the demodulating signal thus closing the signal loop. In a second embodiment, a plurality of frequency locked loop trackers are servoed together centering each one of the trackers on a multiple of the fundamental frequency of the input signal. The resulting phase errors derived from the frequency lock loop trackers are weighted to improve system performance. The frequency corrections from each tracker are weighted with the inverse variance of its tracking performance. Accordingly, harmonics with low variance are weighted strongly, and harmonics in a noisy region of the spectrum and thus high variance will be weighted less strongly. The resulting fundamental frequency estimate is a minimum-variance, and is better than the best signal frequency locked loop estimate. The weighted phase error is then fed back to an integrator to yield a high resolution estimate of the target signal frequency. The amplitude envelopes for each partial signal can be easily extracted and used in conjunction with the fundamental estimate from each frequency lock loop tracker to resynthesize the signal in isolation from the mixture.

Harmonic and frequency-locked loop pitch tracker and sound separation

In a sound (music) synthesis system, a transition between levels of a sound parameter (e.g. volume) is synthesized by fitting a cubic (third degree) spline function between two discontinuous parameter levels. This advantageously eliminates any sound artifacts (pops or clicks) which would otherwise be present due to abrupt changes in the parameter. The cubic spline is fitted (interpolated) to match both the parameter value and its derivative with respect to time at each of the adjacent parameter levels on either side of the transition. The transition thus is advantageously kept brief and in addition it is possible to deal with the situation when the parameter dynamically changes during the transition period, again without causing undesirable sound artifacts.

Method and apparatus for spline parameter transitions in sound synthesis

A discrete-time filter processes digital signals according to a specified function. It has a linear-phase response if all frequencies experience the same time delay when passing through the filter. In this invention computationally efficient digital filters with linear-phase response are constructed according to a very general and computationally efficient algorithm. The invention also specifies a method for constructing fast truncated infinite impulse response filters with polynomial-times-exponential response. The method may be used to design efficient special purpose filters for numerous dedicated applications, or it may be encoded on a chip to permit the creation in real time of many specific filters according to variable filtering parameters. Applications include fast high-resolution frequency estimation (e.g., for pitch tracking and radar) and fast digital filtering in commercial digital audio systems (e.g., for real-time mixing and equalization of signals and digital loudspeaker crossover networks).

Growing community of inventors

Redwood City, CA, United States of America

Avery L Wang