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.

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.

An upper limit of a frequency range of audio indicated by input audio data is detected. A representative point extraction unit downsamples the input audio data to a sampling rate set to be less than or equal to twice the detected upper limit to obtain representative-point audio data. An interpolation processing unit upsamples the representative-point audio data by using a fractal interpolation function (FIF) that uses a mapping function calculated by a mapping function calculation unit, while using the input audio data, if necessary, to generate high-frequency interpolated audio data.

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 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 resampling method based on window function for flexible sampling rate conversion in broadband frequency measurement devices is described. The resampling algorithm can satisfy the requirements of different sampling rates. The frequency responses of the filter in the resampling model based on the Farrow structure are analyzed, and the design criterion of the filter in resampling model is considered. A fractional delay filter design model based on window function method is described. A fractional delay filter matrix, which is expressed by polynomial form, is constructed. Then the expression related to subfilter coefficients is obtained and subfilter coefficients are solved for by the least square method.

A resampling method based on window function for flexible sampling rate conversion in broadband frequency measurement devices is described. The resampling algorithm can satisfy the requirements of different sampling rates. The frequency responses of the filter in the resampling model based on the Farrow structure are analyzed, and the design criterion of the filter in resampling model is considered. A fractional delay filter design model based on window function method is described. A fractional delay filter matrix, which is expressed by polynomial form, is constructed. Then the expression related to subfilter coefficients is obtained and subfilter coefficients are solved for by the least square method.

Embodiments of the present invention provide a digital signal processing apparatus, including an interpolator, an interpolating convolver, or the like, for providing a plurality of output samples or output values in parallel, such as P output samples provided by P Farrow cores, based on a set of input samples or input values, such as 2P+M−2 samples. The digital signal processing apparatus includes a sample distribution logic or structure configured to provide a plurality of subsets of the set of input samples to a plurality of processing cores, such as interpolation cores (e.g., Farrow cores) that perform processing operations associated with different time shifts, for example with respect to a reference time (e.g., a time associated with the input samples). The sample distribution logic includes a hierarchical tree structure having a plurality of hierarchical levels of splitting nodes.

Embodiments of the present invention provide a digital signal processing apparatus including a combiner logic and a plurality of processing cores. Input samples of the digital signal processing apparatus are provided to the plurality of processing cores. Sets of output samples of the processing cores are provided to the combiner logic as input samples, and the sets of samples are provided to the combiner nodes c of the highest hierarchical level (h=0). A digital signal processing apparatus or a parallel decimating digital convolver may be used as a building block of a signal processor application-specific integrated circuit (ASIC) and/or part of other instruments for generating output samples. Furthermore, applications of the digital signal processing apparatus described herein can be addressed on a parallel DSP, in a response time of real-time or near to real-time, for flexible (or almost arbitrary high) sample rates.

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.