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: 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: 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: Circuit systems and methods use prime number interleave optimization for byte lane to time slice conversion of incoming data streams. Generally, the systems and methods buffer data for at least a number of samples equal to the number of byte lanes. Then the samples are transferred to a bank of buffers whose width is the smallest prime number greater than or equal to the number of byte lanes, N. Thus, the systems and methods utilize P minus N phantom lanes. As data is transferred, the data is circularly interleaved (position*N modulo P) so that all data which will be needed at the same time wind up in different readable devices, i.e. the buffers. By appropriate addressing, the data in the different readable devices may then be read in the form of time slices. The process can be reversed for time slice to byte lane conversion.

Abstract: Circuit systems and methods use prime number interleave optimization for byte lane to time slice conversion of incoming data streams. Generally, the systems and methods buffer data for at least a number of samples equal to the number of byte lanes. Then the samples are transferred to a bank of buffers whose width is the smallest prime number greater than or equal to the number of byte lanes, N. Thus, the systems and methods utilize P minus N phantom lanes. As data is transferred, the data is circularly interleaved (position * N modulo P) so that all data which will be needed at the same time wind up in different readable devices, i.e. the buffers. By appropriate addressing, the data in the different readable devices may then be read in the form of time slices. The process can be reversed for time slice to byte lane conversion.

Abstract: A method and apparatus for timing an output signal based on timing of an input signal is provided. A method includes determining a first clock rate derived from the input signal during a first time interval measured by a reference clock. The method also includes signal processing logic to determine a second clock rate during a second time interval based on an error signal that is calculated as a difference between the first clock rate multiplied by the second time interval and a previous value of the second clock rate multiplied by the first time interval.

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: 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 method and apparatus for timing an output signal based on timing of an input signal is provided. A method includes determining a first clock rate derived from the input signal during a first time interval measured by a reference clock. The method also includes signal processing logic to determine a second clock rate during a second time interval based on an error signal that is calculated as a difference between the first clock rate multiplied by the second time interval and a previous value of the second clock rate multiplied by the first time interval.

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 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 system and method for canceling both an echo signal and a spurious sinusoidal signal, such as a telephone dialtone or a carrier signal, from a return signal in a communications system involves producing a replica of the echo signal using an adaptive filter and producing a replica of the sinusoidal signal using an oscillator circuit. The replicas are subtracted from the return signal. The magnitude and phase of the sinusoidal signal are determined by generating a signal that is close in frequency to the expected frequency of the sinusoid and then correlating it to the return signal. The correlation produces a replica of the sinusoid.