Audience, Inc.

Creative Technology Corporation

E-mu Systems, Inc.

Apple Inc.

Knowles Electronics, Inc.

Dana C Massie

David P Rossum

Jean Laroche

Carlo Murgia

Brian D Clark

Michael M Goodwin

Leonardo Rub

Peter Santos

Jean-Marc Jot

Carlos M Avendano

Samuel Charles Dicker

Andrew P Bright

Tom-Davy W Saux

Vladan Bajic

Hanchi Chen

Thomas M Jensen

Luke Dahl

Sumit Sanyal

Victoria Chiu

Vincent Vu

William L Martens

Sol D Friedman

John D Sun

A method for audio signal processing of microphone signals of a headphone. An audio signal from a first microphone of a headphone is filtered by an ANC system to produce a first filtered signal. Dynamic range control is performed upon the first filtered signal to produce a first dynamic range adjusted signal that drives a speaker of the headphone. Other aspects are also described and claimed.

Headphone acoustic noise cancellation and speaker protection or dynamic user experience processing

Systems and methods for real time monitoring of acoustic environment using ultrasound are provided. An example method includes causing production of an acoustic sensing signal. The acoustic sensing signal can be a continuously repeating ultrasonic signal. The method further includes receiving a reflected acoustic signal, the reflected acoustic signal being a representation of an acoustic signal reflected within the acoustic environment. The method further includes analyzing the reflected acoustic signal based at least in part on the acoustic sensing signal to determine properties of the acoustic environment. The acoustic sensing signal can be produced based on a pseudorandom binary sequence. Analyzing the reflected acoustic signal includes correlating the reflected acoustic signal and the acoustic sensing signal to determine a real time estimate of impulse response of the acoustic environment. The method further allows determining that the properties of the acoustic environment have changed.

Real time monitoring of acoustic environments using ultrasound

Arithmetic units and methods for floating point processing are provided. In exemplary embodiments, data paths to and from multiple multipliers and adders are flexibly combined through crossbars and alignment units to allow a wide range of mathematical operations, including affine and SIMD operations. The micro-architecture for a high-performance flexible vector floating point arithmetic unit is provided, which can perform a single-cycle throughput complex multiply-and-accumulate operation, as well as a Fast Fourier Transform (radix-2 decimation-in-time) Butterfly operation.

Multi-function floating point unit

Systems and methods described herein provide for low latency active noise cancellation which alleviate the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques which overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.

Low latency active noise cancellation system

An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Selection of system parameters based on non-acoustic sensor information

An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Especially those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Audio signal bandwidth expansion is performed on a narrow bandwidth signal received from a far end source. The far end source may transmit the signal over the audio communication network. The narrow band signal bandwidth is expanded such that the bandwidth exceeds that of the audio communication network. The signal may be expanded by performing frequency folding on the signal. One or more features are determined for the narrow bandwidth signal, and the expanded signal is modified based on a feature. The feature may be signal band energy slope, narrow band signal energy, or some other feature. The modification may be performed by a shelf filter selected based on the feature. The modified signals are provided for additional processing. In some embodiments, a noise component is added to the narrow band signal prior to folding to create an excitation that reduces the appearance of a fully harmonic signal characteristic.

Low complexity bandwidth expansion of speech

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

Systems and methods described herein provide for low latency active noise cancellation, which alleviates the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques that overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.

Systems and methods for asynchronous sample rate conversion are provided that allow time-varying arbitrary sample ratios. An uncorrected ratio between two arbitrary sample rates is corrected and subsequently used to perform an efficient sample rate conversion on the samples in a data stream. Coefficients of a (polyphase) finite impulse response filter are interpolated based on a current time register value. Additional computational efficiency (and a smaller finite impulse response filter) may be used due to oversampling the input signal to the finite impulse response filter.

Asynchronous sample rate converter

An improved reverberation processor includes a technique for changing environmental parameters without causing disturbing audio artifacts. The technique includes the steps of sequentially changing the read pointers of different delay lines. Additionally, for each delay line a level control variable is ramped down prior to changing the read pointer and then ramped back up. The reverberation processor also provides means for producing and controlling a repeating echo in the reverberation decay, as well as adjusting the diffusion (or echo density) of the reverberation.

Reverberation processor for interactive audio applications

A digital audio spatialization system that incorporates accurate synthesis of three-dimensional audio spatialization cues responsive to a desired simulated location and/or velocity of one or more emitters relative to a sound receiver. Cue synthesis may also simulate the location of one or more reflective surfaces in the receiver's simulated acoustic environment. The cue synthesis techniques are suitable for economical implementation in a personal computer add-on card.

Method and apparatus for three dimensional audio spatialization

A digital audio signal processor for adding harmonic content to an input audio signal through a non-linear transfer function with discontinuities. The discontinuities are generated by bit shifting each input value by an amount dependent on the sign and magnitude of the input value. The amount by which the input value is shifted is roughly inversely related to the magnitude of the logarithm of the input value. The transfer function is fractal and so provides increased harmonic content for all signal amplitudes.

Digital signal processor for adding harmonic content to digital audio

An electronic music system which imitates acoustic instruments addresses the problem wherein the audio spectrum of a a recorded note is entirely shifted in pitch by transposition. The consequence of this is that unnatural formant shifts occur, resulting in the phenomenon known in the industry as 'munchkinization.' The present invention eliminates munchkinization, thus allowing a substantially wider transposition range for a single recording. Also, the present invention allows even shorter recordings to be used for still further memory improvements. An analysis stage separates and stores the formant and excitation components of sounds from an instrument. On playback, either the formant component or the excitation component may be manipulated.

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Capitola, CA, United States of America

Dana C Massie