Abstract: In many audio playback systems, frequencies below a given cut off frequency are suppressed either due to speaker constraints or safety constraints. For example, some speakers are only capable of generating signals above a certain frequency. Prolonged low frequency sound can cause damage to speakers or other components. An audio driver can be equipped with a phantom bass module which by doubling, tripling and/or quadrupling frequencies below a cutoff frequency can simulate the bass experience. The doubling, tripling and quadrupling methods disclosed provide a low complexity formulation of a frequency doubling, tripling and quadrupling. In addition, the frequency doubling, tripling and quadrupling formulations are easily adapted to multi-rate processing, where computational savings can be very high.

Abstract: Traditionally, echo cancellation has employed linear adaptive filters to cancel echoes in a two way communication system. The rate of adaptation is often dynamic and varies over time. Disclosed are novel rates of adaptation that perform well in the presence of background noise, during double talk and with echo path changes. Additionally, the echo or residual echo can further be suppressed with non-linear processing performed using joint frequency-time domain processing.

Abstract: An audio processing system comprising two or more microphones and an echo cancellation system configured to apply a fast converging adaptive filtering algorithm to low frequency bands of a first microphone signal to generate first synthesized echo signal components and an adaptive filtering algorithm to high frequency bands of the first microphone signal to generate second synthesized echo signal components and to apply the first synthesized echo signal components and the second synthesized echo signal components to the first microphone signal to cancel an echo signal of the first microphone signal. An echo estimate and suppression system is configured to receive the first synthesized echo signal components and the second synthesized echo signal components and to apply them to estimate powers of echo signals in one or more additional microphones.

Abstract: A system for processing audio data comprising an amplifier configured to receive an audio signal and to perform nonlinear processing on the audio signal. An encoder coupled to the amplifier, the encoder configured to receive the nonlinearly processed audio signal and to encode the nonlinearly processed audio signal into a data transmission format. A transmitter configured to receive and transmit the encoded nonlinearly processed audio signal. A receiver configured to receive the transmitted encoded nonlinearly processed audio signal and to decode the encoded nonlinearly processed audio signal. A digital voice processor configured to receive the nonlinearly processed audio signal and to use the nonlinearly processed audio signal for echo estimation and to subsequently subtract the estimated echo signal from a microphone signal.

Abstract: A system for detecting motion comprising a first speaker, a first microphone separated from the first speaker by a distance D1, a sound generator, an echo parameter measurement device and an echo parameter monitor, wherein the echo parameter monitor stores two or more sequential echo parameters and generates a motion indicator if the two or more sequential echo parameters indicate a change in an acoustic echo path that exceeds a predetermined threshold.

Abstract: A system for controlling distortion comprising a total harmonic distortion (THD) modeling system configured to apply a chirp signal to a system and to identify one or more frequency bands at which distortion is present, and to apply a ramping signal to identify for each of the one or more frequency bands an input signal level at which distortion is initiated, a signal processing system configured to receive an input signal, to determine whether frequency components are present in the input signal that are associated with the one or more frequency bands at which distortion is present, and to limit the amplitude of the input signal at the one or more frequency bands, such as by applying dynamic range compensation.

Abstract: A system and method for defringing chromatic aberrations that occur in imaging devices such as scanners. The system comprises shift filters to shift lines in the various color planes together. In addition in each color plane, a spread filter is used to compensate for the unequal point spread functions of each color. Furthermore, the results can be enhanced by filtering in the luminance-chrominance space.

Abstract: An apparatus for abstract memory addressing. A processor for generating an abstract memory address. A base register for storing a base memory address. An adder for adding the base memory address to the abstract memory address and generating a physical address for a device memory. A pointer register for storing the physical address, wherein the pointer register is directly coupled to the device memory.

Abstract: Poly-phase filters are used to offer an efficient and low complexity solution to rate conversion. However, they suffer from inflexibility and are not easily reconfigured. A novel design for rate converters employ poly-phase filters but utilize interpolation between filter coefficients to add flexibility to rate conversion. This interpolation can be implemented as an interpolation of the poly-phase filter results. Additional approximations can be made to further reduce the amount of calculations required to implement a flexible rate converter.

Abstract: A system for improved audio in a headset comprising a first headset microphone generating a first signal. A second headset microphone generating a second signal. A multiplexer coupled to the first headset microphone and the second headset microphone for multiplexing the first signal and the second signal. A power extractor for extracting power for use by one or more of the multiplexer, the first headset microphone and the second headset microphone. A demultiplexer for extracting the first signal and the second signal. A signal processor for generating a noise reduced microphone signal. An audio subsystem for receiving the noise reduced microphone signal and for generating speaker signals for a first headphone speaker and a second headphone speaker.

Abstract: An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.

Abstract: Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency band.

Abstract: Poly-phase filters are used to offer an efficient and low complexity solution to rate conversion. However, they suffer from inflexibility and are not easily reconfigured. A novel design for rate converters employ poly-phase filters but utilize interpolation between filter coefficients to add flexibility to rate conversion. This interpolation can be implemented as an interpolation of the poly-phase filter results. Additional approximations can be made to further reduce the amount of calculations required to implement a flexible rate converter.

Abstract: An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.

Abstract: A system for performing floating point operations comprising a floating point multiply function that utilizes one or more fixed point functional blocks of a processor and one or more dedicated floating point functional blocks of the processor. A floating point add function that utilizes one or more fixed point functional blocks of a processor and one or more dedicated floating point functional blocks of the processor. A floating point normalize function that utilizes one or more fixed point functional blocks of a processor and one or more dedicated floating point functional blocks of the processor.

Abstract: Acoustic echoes in communications systems are distracting and undesirable. Acoustic echoes occur in communications systems where sound produced by a speaker is picked up by a microphone in a communications system. In a stereo playback environment, echo cancellation techniques become more complicated. Echo cancellation can be performed by performing echo cancellation on a center signal, which is the sum of a left channel signal and the right channel signal, or left signal and a difference signal, which is the difference of the right channel signal and the left channel signal. The adaptation rates of the two echo cancellers meet certain constraints to prevent degeneracies in the echo cancellation system.

Abstract: A system and apparatus for constructing a displacement model across a frequency range for a loudspeaker is disclosed. The resultant displacement model is centered around the distortion point. Once a distortion model is constructed it can be incorporated into an audio driver to prevent distortion by incorporating the model and a distortion compensation unit with a conventional audio driver. Various topologies can be used to incorporate a distortion model and distortion compensation unit into an audio driver. Furthermore, a wide variety of distortion compensation techniques can be employed to avoid distortion in such an audio driver.

Abstract: A system and method for defringing chromatic aberrations that occur in imaging devices such as scanners. The system comprises shift filters to shift lines in the various color planes together. In addition in each color plane, a spread filter is used to compensate for the unequal point spread functions of each color. Furthermore, the results can be enhanced by filtering in the luminance-chrominance space.

Abstract: Peak reduction and power limitations are used to prevent distortion and protect components. In a cellular telephone, peak reduction can be based on battery power level to prevent electrical distortion from saturation. In addition peak reduction can be used to prevent mechanical distortion such as rub and buzz. Dynamic range compression can be used for peak reduction. In another application dynamic range compression can be used to control the power output to protect a speaker from damage. One example of a dynamic range compressor/peak limiter comprises a look-ahead buffer and an analysis engine. For example, the look-ahead buffer holds a window of samples of a signal. The analysis engine selects a gain envelope function on the basis of the samples, for example, by selecting the Pth sample in the buffer whenever that sample exceeds a given threshold.

Abstract: In many audio playback systems, frequencies below a given cut off frequency are suppressed either due to speaker constraints or safety constraints. For example, some speakers are only capable of generating signals above a certain frequency. Prolonged low frequency sound can cause damage to speakers or other components. An audio driver can be equipped with a phantom bass module which by doubling, tripling and/or quadrupling frequencies below a cutoff frequency can simulate the bass experience. The doubling, tripling and quadrupling methods disclosed provide a low complexity formulation of a frequency doubling, tripling and quadrupling. In addition, the frequency doubling, tripling and quadrupling formulations are easily adapted to multi-rate processing, where computational savings can be very high.