Abstract: 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.

Abstract: An aspect of the present disclosure is directed to systems, methods, and software products that enhance the sound and perception of bass energy in loudspeaker systems which have limited response in the low frequency spectrum. The systems, methods, and software products provide for (i) receiving one or more bass sub-bands of audio channel content, wherein each bass sub-band is separated from an audio program content spectrum including high-band energy; (ii) applying boost to each bass sub-band to compensate for rolloff of the speaker system; (iii) applying a limiter to prevent speaker saturation at low frequencies and/or high volume levels; and (iv) combining each bass sub-band along with the high band energy for sending as a recombined audio signal to the speaker system for rendition as physical sound.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A disclosed system and method dynamically controls the perceived volume of a stereo audio program including left and right channel signals. The system includes a dynamic volume control that operatesto maintain a perceived constant volume level of the stereo audio program; and an excessive spatial processing protection processor configured and arranged for controlling the level of a difference signal, created as a function of the right channel signal subtracted from the left channel signal (L?R), relative to the level of a sum signal, created as a function of the right channel signal plus the left channel signal. The excessive spatial processing protection processor processes the audio signals so as to control the difference (L?R) signal relative to the sum (L+R) signal. A system and method are also provided for dynamically controlling the perceived volume of a stereo audio program including left and right channel signals.

Abstract: Systems and methods for enhancing the low frequency response of a loudspeaker for relatively low input level audio signals and protect the loudspeaker for relatively high input level audio signals make use of a crossover network configured so separate an audio input signal into at least two frequency bands including a low frequency band; and a signal compressor responsive to the energy level of the low frequency portion of an input audio signal in the low frequency band and configured to provide amplification gain on the low frequency portion of the input signal when the energy level of the low frequency portion of the input signal is relatively low. One or more peak limiters may be positioned before the summer(s), in the circuit, but after the low band compressor. The EQ unit(s) may be positioned before all processing to the L and R inputs.

Abstract: An integrated circuit (IC) includes a programmable gain amplifier. The programmable gain amplifier comprises a first-stage amplifier configured to operate with at least one relatively high power supply voltage in order to accommodate at the input of the first-stage amplifier a relatively large range of input signals, the first-stage amplifier having a gain setting that is adjustable from a set of predetermined gain settings separated in relatively coarse increments so as to minimize the number of analog switches that must be implemented with high-voltage active devices in order to set each gain setting.

Abstract: 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.

Abstract: The disclosed BTSC encoder includes a left high pass filter means for receiving a digital left channel audio signal and for digitally high pass filtering the digital left channel audio signal and thereby generating a digital left filtered signal; a right high pass filter means for receiving a digital right channel audio signal and for digitally high pass filtering the digital right channel audio signal and thereby generating a digital right filtered signal; a matrix means for receiving the digital left and digital right filtered signals, and including means for summing the digital left and digital right filtered signals and thereby generating a digital sum signal, and including means for subtracting one of the digital left and digital right filtered signals from the other of the digital left and digital right filtered signals and thereby generating a digital difference signal; a difference channel processing means for digitally processing the digital difference signal; and a sum channel processing means for d

Abstract: A reconfigurable network arrangement of resistors and switches is constructed so that it can be coupled to one or more operational amplifiers and selectively programmed so as to set the gain of the resulting amplifier. The configuration of the network arrangement of resistors and switches to include resistors that can be connected in the feedback path in series and in parallel with each other is such as to provide a wider selection of gain settings, without the need to increase the physical area of the switches on a integrated circuit arrangement.

Abstract: A switching circuit for switching a time-varying input signal, the switching circuit comprising: at least one switch including a N-channel MOSFET and a P-channel MOSFET, each having a gate configured to receive a drive signal to change the ON/OFF state of the switch; and a drive circuit configured and arranged so as to selectively apply a pair of drive signals to change the ON/OFF state of the switch, the drive circuit being configured and arranged to generate the drive signals as a function of (a) a pair DC signal components sufficient to change the ON/OFF state of the switch and (b) a pair of time-varying signal components as at least a partial replica of the signal present on the source terminal of each MOSFET so that when applied with the DC signals to the gates of the re-channel MOSFET and p-channel MOSFET respectively, the drive signals will be at the appropriate level to maintain the ON/OFF state of the switch and keep the gate-source voltages of each MOSFET within the gate-source breakdown limit of th

Abstract: An integrated circuit (IC) includes a programmable gain amplifier. The programmable gain amplifier comprises a first-stage amplifier configured to operate with at least one relatively high power supply voltage in order to accommodate at the input of the first-stage amplifier a relatively large range of input signals, the first-stage amplifier having a gain setting that is adjustable from a set of predetermined gain settings separated in relatively coarse increments so as to minimize the number of analog switches that must be implemented with high-voltage active devices in order to set each gain setting.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A disclosed system and method dynamically controls the perceived volume of a stereo audio program including left and right channel signals. The system comprises: a dynamic volume control configured and arranged so as to maintain a perceived constant volume level of the stereo audio program; and an excessive spatial processing protection processor configured and arranged for controlling the level of a difference signal, created as a function of the right channel signal subtracted from the left channel signal (L?R), relative to the level of a sum signal, created as a function of the right channel signal plus the left channel signal;; wherein the excessive spatial processing protection processor processes the audio signals so as to control the difference (L?R) signal relative to the sum (L+R) signal.