That Corporation

Matthew Easley

Matthew S Barnhill

Roger R Darr

W Kevin Kay

Mikio Oda

William Kevin Kay

Timothy Brault

Fen Xu

Miko Oda

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

Frequency domain compensation is provided for spectral impairment resulting from the audio path characteristics for a given audio device in a given listening space. Selected segments of an audio stream are recorded at a listener position to measure degradation in the audio path and to update compensation filter characteristics of the audio device. Recorded transmitted and received audio sequences are aligned based and compared in the frequency domain. The difference between the aligned transmitted and received sequences represents the frequency domain degradation along the acoustic path due to the speaker, the physical attributes of the room, and noise. A dynamically updated noise model is determined for adjusting compensation filter characteristics of the audio device, which can be updated during use of the audio device. A compensation curve is derived which can adapt the equalization of the audio device passively during normal usage.

Passive sub-audible room path learning with noise modeling

Systems are disclosed including a television (TV) set having included audio system. The TV set permits control over various functions, at least including audio volume, via a remote control. When the viewer activates the remote volume control, a graphic appears indicating the state of the volume control and, optionally the mute status. The graphic can be presented on the TV screen. In alternate embodiments, the graphic may be presented on a display of the remote, or even some other location, e.g., a different remote control. If a mute button is provided on the remote control, when the viewer activates the mute button, a graphic appears indicating the state of muting and, optionally, the volume control status. The TV set also offers control over various aspects of the audio system, including settings which go beyond volume up/down, generally through some sort of menu system.

System for configuration and status reporting of audio processing in TV sets

Audio processing architectures are described for implementing a multi-band audio compression algorithm with advanced surround processing. Exemplary architectures can improve the fidelity and perceived sound field spread of inexpensive, cabinet mounted, stereo speakers such as those that might be found in televisions, wireless speaker systems and sound bars. Embodiments of the present disclosure can improve inexpensive, cabinet mounted, stereo speakers by providing, e.g., (i) an Advanced Surround algorithm that adds depth and height to the left/right/center sound field images, (ii) a Soft Clip algorithm to minimize the perceived artifacts caused by compressor overshoot, (iii) configurable crossover filter order adjustment to allow better isolation between bands, (iv) a compressor maximum gain adjustment to reduce overshoot and minimize noise boost, and/or (v) a center gain adjustment to emphasize the perception of the center image (dialog) in high ambient noise situations.

Configurable multi-band compressor architecture with advanced surround processing

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 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

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.

Growing community of inventors

Woodstock, GA, United States of America

Matthew Easley