A method for providing back-compatibility for rational sampling rate disparities between two circuitries, comprises: a) providing a Phase Locked Loop (PLL) operating at a rate different than that of the Symbols generator, which is coupled to a Digital to Analog Converter (DAC) or an Analog to Digital Converter (ADC); b) providing an interpolation filter coupled to said converter, which filter is adapted to perform sampling rate conversion operations on the samples using zero-stuffing, filtering, and decimation, or the like computation-saving algorithm; and c) obtaining the sampling of the symbols at the required and compatible rate.

A variable output data rate converter circuit preferably meets performance requirements while keeping the circuit complexity low. In some embodiments, the converter circuit may include an oversampling sigma delta modulator circuit to quantize an analog input signal at an oversampled rate, and output an sigma delta modulated signal, a transposed polynomial decimator circuit to decimate the sigma delta modulated signal, and output a first decimated signal, and an integer decimator circuit to decimate the first decimated signal by an integer factor and output a second decimated signal having a desired output data rate. The transposed polynomial decimator circuit has a transposed polynomial filter circuit and a digital phase locked loop circuit, which tracks a ratio between a sampling rate of the first decimated signal and the oversampled rate, and outputs an intersample position parameter to the transposed polynomial filter circuit.

A method, a computer-readable medium, and an apparatus for resampling audio signal are provided. The apparatus resamples the audio signal in order to preserve the audio playback quality when dealing with audio playback overrun and underrun problem. The apparatus may receive a data block of the audio signal including a first number of samples. For each sample of the first number of samples, the apparatus may slice a portion of the audio signal corresponding to the sample into a particular number of sub-samples. The apparatus may resample the data block of the audio signal into a second number of samples based on the first number of samples and the particular number of sub-samples associated with each sample of the first number of samples. The apparatus may play back the resampled data block of the audio signal via an electroacoustic device.

Various exemplary embodiments are directed to methods including obtaining an input sample magnitude, filtering the obtained input sample magnitude, generating a sample-to-sample difference based on the filtered input sample magnitude, and engaging an actuator in accordance with a determination that the sample-to-sample difference satisfies a rate threshold. In addition, various exemplary embodiments are directed to devices including a processor, a control sensor operatively coupled to the processor and operable to obtain an input sample magnitude, an input filter operatively coupled to the processor and operable to filter the at least one obtained input magnitude sample, a non-transitory computer-readable medium operatively coupled to the processor and including a rate engine operable to generate a sample-to-sample difference based on the filtered input sample magnitude, and to generate a determination that the sample-to-sample difference satisfies a rate threshold, and a control actuator operatively coupled to the processor and operable to engage an operation mechanism in accordance with the determination that the sample-to-sample difference satisfies a rate threshold.

Systems and methods for low power lattice wave filters include an input operable to receive a digital input signal having a first sample rate, a first processing branch including a first delay element operable to receive the digital input signal and output a delayed digital input signal, a second processing branch including a first adder operable to receive the digital input signal and subtract a delayed feedback signal to produce a difference signal, a second adder operable to combine the delayed digital input signal and the difference signal to produce an output signal, and wherein the second processing branch further includes a feedback path including a second delay element operable to receive the output signal and output the delayed feedback signal. In a multistage topology, a register is disposed between each stage and clocked to reduce ripple power.

Systems and methods for low power lattice wave filters include an input operable to receive a digital input signal having a first sample rate, a first processing branch including a first delay element operable to receive the digital input signal and output a delayed digital input signal, a second processing branch including a first adder operable to receive the digital input signal and subtract a delayed feedback signal to produce a difference signal, a second adder operable to combine the delayed digital input signal and the difference signal to produce an output signal, and wherein the second processing branch further includes a feedback path including a second delay element operable to receive the output signal and output the delayed feedback signal. In a multistage topology, a register is disposed between each stage and clocked to reduce ripple power.

Disclosed herein is a method and apparatus for converting a stream of samples at a first sampling rate to a stream of samples at a second sampling rate. An exemplary method includes measuring the first sampling rate; determining a first upsampling factor from a basis including: the measured first sampling rate, the target value of the second sampling rate, and a resynchronisation error factor, the first upsampling factor being constrained to be an integer power of a predetermined integer value; and deriving, from a reference set of filter coefficients and from a ratio of the first upsampling factor to a reference upsampling factor, a first set of filter coefficients for use in a first interpolation filter, the reference set of filter coefficients being for a reference upsampling factor that is an integer power of the predetermined integer value.

A signal conversion from an input signal to an output signal where the filter used is factorized so that the conversion comprises determining 1) only a first factor at each sampling time of the input signal, where this first factor is independent on the sampling times of the output signal, and 2) only a second factor at each sampling time of the output signal, where this second factor is independent of the sampling times of the input signal. This reduces the computational load for this conversion. In addition, for most filters, the factors may be calculated recursively further increasing the computational load and also reducing the storage requirements. This allows for instantaneous changes in the sampling rates or non-uniform sampling rates with low computational requirements and low memory usage.

A method of converting a stream of samples at a first sampling rate to a stream of samples at a second sampling rate is disclosed, comprising: measuring the first sampling rate; determining a first upsampling factor from a basis comprising: the measured first sampling rate, the target value of the second sampling rate, and a resynchronisation error factor, the first upsampling factor being constrained to be an integer power of a predetermined integer value; and deriving, from a reference set of filter coefficients and from a ratio of the first upsampling factor to a reference upsampling factor, a first set of filter coefficients for use in a first interpolation filter, the reference set of filter coefficients being for a reference upsampling factor that is an integer power of the predetermined integer value.

A device for processing a digital signal includes a Farrow structure (14) that applies to the digital signal a time-varying sample rate conversion from the fixed sample rate to a time varying sampling. The digital signal sampled at the time varying sampling is a resulting signal. The Farrow structure (14) is controlled from a control variable. A spectral analysis means (15) performs a spectral analysis of the resulting signal to determine the frequency values of the resulting signal. A determining means (16) determines a sparseness parameter of the frequency values of the resulting signal. A controlling means (17) modifies the control variable according to the value of the sparseness parameter.