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.

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, devices, and methods related to a sample rate converter (SRC) for implementing a rate conversion R are provided. The SRC receives input samples at an input rate F.sub.in and outputs samples at an output rate F.sub.out=F.sub.in×R, where R is a fractional value greater than 1. The SRC includes a plurality of filters to process the received input samples and a multiplier-adder block to generate the output samples based on respective delta values and outputs of the plurality of filters. The SRC further includes a plurality of buffers to buffer samples between the plurality of filters and the multiplier-adder block based at least in part on N buffer read pointers, where N is an integer greater than 1. The SRC further includes resampler control circuitry to generate N delta values of the delta values and the N buffer read pointers in parallel based on R.

Provided, among other things, is an apparatus for digitally processing a discrete-time signal that includes: an input line for accepting an input signal, processing branches coupled to the input line, and an adder coupled to outputs of the processing branches. First and second lowpass filters, each having a frequency response with a magnitude that varies approximately with frequency according to a product of raised functions, are included within baseband processors in such processing branches.

Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.

Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.

A measuring arrangement acquires signals of alternating electrical magnitudes. A sampling apparatus performs a sampling of the signals to form digital sample values. A clock tracking apparatus adapts a sampling clock used by the sampling apparatus in the light of the frequency of the signal to be sampled. In order to be able to acquire reliably signals of alternating electrical magnitudes even when they have different frequencies, the sampling apparatus samples at least two of the signals each with its own sampling clock and the clock tracking apparatus adapts the sampling clock in the light of the frequency of the signal to be sampled simultaneously for each of these at least two signals. There is also described a corresponding method for measuring electrical signals.

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.

The present application discloses a filtering method and a filtering device of a filter, a filter and a storage medium. The method includes: obtaining a clock input signal and a clock output signal and comparing them, and determining a phase relationship between the clock input signal and the clock output signal according to a comparison result; determining a decimal deviation factor according to the phase relationship in determining that the phase relationship meets a preset condition; and filtering a sample input signal according to the decimal deviation factor to obtain a filtered sample output signal. The present application can obtain an accurate decimal deviation factor, by obtaining the phase relationship between the clock input signal and the clock output signal, in determining that the phase relationship meets a preset condition, and can adjust the sample input signal according to the decimal deviation factor to obtain a smooth sample output signal.