That Corporation

Other

Matthew S Barnhill

Matthew Easley

Roger R Darr

W Kevin Kay

William Kevin Kay

Mathew F Easley

Systems, methods, and algorithms for estimating the sample clock difference between an audio playback and recording device are described. The sample clock difference can be determined by sequentially playing the same pseudorandom signal and exploiting the statistical properties of said signal in the recording. An exemplary embodiment utilizes a maximum length sequence (MLS) for the pseudorandom signal. While an MLS sequence is a good choice for clock synchronization, it is possible to use other pseudorandom sequences. The sequence preferably has a sharp correlation peak that is clearly discernable with even a single sample shift and be of sufficient length to support the anticipated clock drift.

Audio sampling clock synchronization

Systems and methods utilize a modified genetic algorithm for adapting an off-the-shelf audio system, such as in a high-end television, to a given, particular room or other physical location presenting a specific or unique auditory environment with a set of acoustic properties. An audio system is adapted to a given room by determining an IIR based EQ solution via iterative techniques, including an iterative technique based upon a genetic algorithm adapted for an audio frequency response equalization application. In a variant, an audio system is adapted to a particular room, adjust the EQ across a microphone's bandwidth while preserving the factory-calibrated EQ response across the remaining bandwidth.

Genetic-algorithm-based equalization using IIR filters

A NICAM audio signal re-sampler may include a non-linear interpolator configured to interpolate in a non-linear manner between sequential digital samples that are based on a stream of demodulated NICAM audio samples. A phase differential calculator may be included that compares phase information at different resolutions.

NICAM decoder with output resampler

A television audio signal encoder includes a device that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The device also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoder also includes a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal. The set of filter coefficients is applied to the difference signal by a single multiplier in a recursive manner to prepare the difference signal for transmission.

Configurable recursive digital filter for processing television audio signals

Television audio signal encoders include a matrix that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The matrix also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoders also include a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal and/or the sum signals. Each selectable set of filter coefficients is associated with a unique filtering application to prepare the difference signal for transmission. Decoders with the matrix are also disclosed.

Configurable filter for processing television audio signals

A digital audio encoder, digital video conditioner, and a digital modulator are described for producing a television broadcast signal at a desired channel frequency range. Left and right audio channel signals are digitized and encoded according to a stereo standard and then combined to form a stereo audio signal. A second audio programming channel signal may be included. A video input can be digitized and conditioned to form a digital video channel. The stereo audio signal can be placed directly at a desired channel frequency by frequency modulation without the need for using an intermediate frequency. The digital video channel can be placed at a desired frequency by amplitude modulation. The digital and audio channels can be digitally combined to create a television transmission signal at a desired frequency and according to a desired standard.

Direct digital encoding and radio frequency modulation for broadcast television applications

NICAM audio signal resampler

A signal quality estimation and control system for controlling a signal having noise. The system may include a signal strength determination system having an input configured to receive the signal and an output configured to output information indicative of the strength of the signal, a noise strength determination system having an input configured to receive the signal and an output configured to output information indicative of the strength of the noise in the signal, and a signal quality evaluation system. The signal quality evaluation system may have an input configured to receive the output of the signal strength determination system, an input configured to receive the output from the noise strength determination system, and an output configured to output information indicative of the quality of the signal based on the output from the signal strength determination system and the output from the noise strength determination system. A signal control system may be include that has an input configured to receive the output from the signal quality evaluation system and to control a switching function relating to the signal based on the output from the signal quality evaluation system.

Signal quality estimation and control system

An algorithm calculates spectral compression/expansion filter coefficients using a value proportional to a reciprocal of a feedback/feedforward signal and stores the coefficients to a lookup table. The lookup table is indexed by a pre selected set of coefficient bits to generate a filter coefficient function. A first portion of the lookup table stores a plurality of discrete values at index points of a line segment corresponding to a filter coefficient function approximation, so as to generate an initial discrete value corresponding to the filter coefficient function at a value of the high order bits and a second portion stores a plurality of slope values, which indicates a slope of the filter coefficient function. A linear circuit interpolates/decimates an approximation of the compression/expansion filter coefficient function based on the slope value, the initial discrete value and a preselected set of low order bits of the reciprocal value.

Reciprocal index lookup for BTSC compatible coefficients

A television audio signal encoder includes an up-sampler that inserts additional samples into a sum signal, a difference signal, and/or a secondary audio program signal to increase the sample rate of the corresponding signal or signals. The sum signal represents the sum of a left channel digital audio signal and a right channel digital audio signal and the difference signal represents the difference between the left and right audio signals.

Up-sampling television audio signals for encoding

A television audio signal encoder includes a matrix that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The matrix also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoder also includes a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal. Each selectable set of filter coefficients is associated with a unique filtering application to prepare the difference signal for transmission.

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Duluth, GA, United States of America

Matthew S Barnhill