Creative Technology Corporation

E-mu Systems, Inc.

Knowles Electronics, Inc.

Audience, Inc.

Rossum Electro-music, LLC

Creative Labs, Inc.

Other

Knowler Electronics, LLC

David P Rossum

Niel David Warren

Dana C Massie

Jean Laroche

Henrik Thomsen

Kim Spetzler Berthelsen

Dibyendu Nandy

Robert A Popper

Andrzej Pawlowski

Ramanujapuram Raghuvir

Yang Li

Wong Hoo Sim

Thomas Craig Savell

Brian D Clark

Jay Scott Fuller

Leonardo Rub

Ronald Neal Dow

Mark Barry Dolson

Erick Alvarado

Matthew F Williams

Sneha Date

Xiaojun Chen

Donald F Ruffcorn

Robert S Crawford

Michael Guzewicz

William L Martens

Sol D Friedman

John D Sun

Khir Hien Tan

An audio filter and corresponding method with through-zero linearly variable resonant frequency are described. An example method includes receiving an input signal; receiving one or more control input signals; and providing a resonant filter for electronic music for processing the input signal. The resonant filter can include an exponential times linear multiplication to produce the resonant frequency control responsive to control voltages exponentially for musical octave/semitone control, and linearly for timbre modulation, the waveform being invariant over corresponding exponential changes in pitch of both input signal and filter resonance. The linear modulation can include inverting the sign of a bandpass feedback based on the sign of the resonant frequency variable so as to maintain stability. In the example method, to prevent a glitch on the output when frequency changes sign, a separate highpass output signal may be generated.

Audio filter with through-zero linearly variable resonant frequency

Methods and systems for generating oscillatory timbres for musical synthesis through synchronous ring modulation are described. An example method performing hard synchronization comprising using first and second oscillators, the first oscillator being a fundamental oscillator which provides a fundamental frequency, the second oscillator being a modulation oscillator operable at a frequencies higher than the first oscillator; and in response to the fundamental oscillator completing its cycle, synchronizing the modulation oscillator to the original point of its waveform. Ring modulation may be performed on the synchronized output by multiplying it by a waveform of the fundamental oscillator, such that the ring modulation is synchronized to produce a variety of oscillatory timbres. The method can create a variety of unique sounds having musically pleasing characteristics. A real-time audio signal can be used instead of the first oscillator to produce dynamically varying timbres.

Oscillatory timbres for musical synthesis through synchronous ring modulation

A multi-core audio processor includes a plurality of data ports configured to receive and transmit data, wherein a first data port of the plurality of data ports is configured to receive data associated with a first stream, a first plurality of audio interface resources configured to perform operations on the data, and a multiplexing unit configured to route data between the plurality of data ports and the first plurality of audio interface resources, the multiplexing unit including a multiplexer associated with each of the first plurality of audio interface resources. The multi-core audio processor also includes a first plurality of transport buffers, at least one transport buffer associated with each of the first plurality of audio interface resources. A first set of buffers of the first plurality of transport buffers is configured to synchronously store data received from the first set of the first plurality of audio interface resources.

Multi-core audio processor with phase coherency

The present disclosure provides a multi-channel morphing digital audio filter, including methods and systems for real-time interpolation between a plurality of fixed frequency responses. Real-time interpolation may be performed simultaneously along three separate axes. Various embodiments of the present technology include receiving, in real-time via a digital filter module, at least one acoustic signal and one or more control input signals indicative of one or more corresponding interpolator values. The methods and systems may further include determining one or more filter coefficients based on a linear interpolation in an encoded space of a plurality of fixed frequency responses, generating a filter from the one or more determined filter coefficients, and applying the generated filter to the at least one received acoustic signal. In some embodiments, frequency and amplitude (i.e., angle and radius) of poles and zeros of the filters may be processed independently, and poles may be unconditionally stable.

Multi-channel morphing digital audio filter

Provided are systems and methods for utilizing digital microphones in low power keyword detection and noise suppression. An example method includes receiving a first acoustic signal representing at least one sound captured by a digital microphone. The first acoustic signal includes buffered data transmitted with a first clock frequency. The digital microphone may provide voice activity detection. The example method also includes receiving at least one second acoustic signal representing the at least one sound captured by a second microphone, the at least one second acoustic signal including real-time data. The first and second acoustic signals are provided to an audio processing system which may include noise suppression and keyword detection. The buffered portion may be sent with a higher, second clock frequency to eliminate a delay of the first acoustic signal from the second acoustic signal. Providing the signals may also include delaying the second acoustic signal.

Utilizing digital microphones for low power keyword detection and noise suppression

At a microphone, voice activity is detected in a data stream while simultaneously buffering audio data from the data stream to create buffered data. A signal is sent to a host indicating the positive detection of voice activity in the data stream. When an external clock signal is received from the host, the internal operation of the microphone is synchronized with the external clock signal. Buffered data stream is selectively sent through a first path, the first path including a buffer having a buffer delay time representing the time the first data stream takes to move through the buffer. The data stream is continuously sent through a second path as a real-time data stream, the second path not including the buffer, the real-time data stream beginning with the extended buffer data at a given instant in time. The buffered data stream and the real-time data stream are multiplexed onto a single data line and transmitting the multiplexed data stream to the host.

Microphone apparatus and method with catch-up buffer

A new and more efficient filtering system (e.g., digital microphone decimation filter architecture system) is described. A key to this architecture is the use of two parallel filter paths. Each path operates at the output sample rate, and comprises a shorter FIR filter followed by a series of allpass stages (e.g., implementing IIR filters). The FIR filter is designed to remove all but the last octave of out-of-band noise. The allpass stages are designed such that when the two paths are summed together, the out-of-band noise for the final octave cancels out, leaving only the desired sign.

Efficient digital microphone decimation filter architecture

Provided are methods and systems for reducing a transition rate in transmitting data between analog and digital chips in Sigma-Delta Modulator (SDM) based Digital to Analog Converters (DACs) and Analog to Digital Converters (ADCs) intended to be used in audio signal processing. An example method may comprise receiving, by a digital chip, SDM binary data, mapping the SDM binary data to transition binary codes, and transmitting the transition binary codes to an analog chip. The mapping can be carried out according to a principle that the more commonly used SDM binary data codes correspond to transition binary data codes that require that fewer transitions occur in the signals between the chips. The methods and systems described provide for lowering the power needed for carrying out the data transmission between digital and analog chips.

Reducing data transition rates between analog and digital chips

Systems and methods for sample rate tracking are provided. An example method includes computing an actual latency associated with an output sample from an output sample stream. The actual latency is calculated using a phase and a phase increment (conversion rate ratio). A measured latency is determined using an internal clock using a presentation time of the output sample, or an input sample from an input sample stream, or both. The measured latency is compared to the actual latency to generate a latency error. A successive phase increment can be determined based on the latency error by using a low-pass or adaptive filter to adjust the latency error.

Systems and methods for sample rate tracking

Digital methods and systems for signal processing and filtering are provided. The methods and corresponding systems provide asynchronous conversion of sampling rate frequencies and utilize advanced multistage phasor filters for converting an input signal having a first sampling rate into an output signal sampled in an arbitrary sequence of sampling times. The conversion process provides a sequence of sets of complex numbers representing a filtered version of the input signal. More specifically, the conversion process includes the calculation of values of the output signal by multiplying (e.g., scaling) the sets of complex numbers by a corresponding set of complex phasors, the complex phasors corresponding to the timing of the arbitrary time sequence to obtain a corresponding set of real results with the value of the output signal being the sum of the real results.

Sample rate conversion using infinite impulse response filters

A method and a computer program product for sample rate conversion that features distributive or hybrid filtering to reduce unwanted artifacts, such as aliasing and the computational requirements to avoid the aforementioned artifacts. The method includes receiving, at a first sample rate, a plurality of data points, associated with a first signal, operating on the plurality of data points to associate the signal with a predetermined set of parameters, with the set of parameters including a first transition band having an image associated therewith; and varying the sample rate associated with the first signal by interpolation with an interpolator having associated therewith a second transition band, with the width associated with the second transition band being a function of a spectral separation between the first transition band and its image, wherein a second signal is produced having a sequence of data samples approximating the first signal.

Sample rate converter having distributed filtering

A method and apparatus for printing a one-dimensional, continuous signal, such as a sound wave, as a printed pattern and for recovering the sound wave from the printed pattern. The method decomposes the one dimensional signal two-dimensional function that is continuous in both dimensions. The two-dimensional function can be printed as a pattern on to a paper substrate using a standard printer. The audio signal is recovered by scanning the pattern and processing the scanned values.

Method of transmitting a one-dimensional signal using a two-dimensional analog medium

A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory.

Growing community of inventors

Aptos, CA, United States of America

David P Rossum