One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

Systems and method for resampling are provided. A method of resampling includes receiving a first sampled signal that is sampled at a first sample rate, where the first sample rate is a submultiple of a system clock rate for a Farrow filter. The method further includes resampling the first sampled signal, using the Farrow filter having a plurality of finite impulse response (FIR) filters and an arbitrary position interpolator, at a second sample rate to generate a second sampled signal. The interpolation factor for each sample of the second sampled signal is retrieved from at least one lookup table stored in memory and the first sample rate and the second sample rate are fixed and locked to a common frequency reference. The method further includes outputting the second sampled signal at the second sample rate.

Filters are discussed where a first window function and a second window function are applied to a digital input signal, wherein a window length of the first window function is longer than a window length of the second window function. The results of this windowing are integrated.

Aspect of the present disclosure provide for a circuit. In an example, the circuit comprises a multiplexer having a first input, a second input, a control input, and an output. The circuit further comprises a first register having an input coupled to the output of the multiplexer and an output. The circuit further comprises a second register having an input coupled to the output of the first register and an output. The circuit further comprises a subtractor having a first input coupled to the output of the multiplexer and a second input coupled to the output of the second register. The circuit further comprises a third register having an input coupled to the output of the subtractor and an output coupled to the first input of the multiplexer.

A process for processing a digital signal comprises constructing a fractional order control system that models a desired frequency response by assembling filter components from a filter component library. The filter components are defined by Laplace functions that include a non-integer control order having a variable fractional scaling exponent. Then, the fractional order control system is adjusted by applying an altitude exponent to the fractional order control system, and the altitude exponent changes a magnitude of the frequency response without changing a width of a transition band of the frequency response. An input signal in the digital frequency domain is received and processed based upon the fractional order control system to generate a digital output that is conveyed.

Filters are discussed where a first window function and a second window function are applied to a digital input signal, wherein a window length of the first window function is longer than a window length of the second window function. The results of this windowing are integrated.

Deterioration in audio quality is inhibited in a device which records audio and stretches a reproduction time period. A sampling processing unit performs processing of sampling audio in a predetermined period at a sampling rate higher than a predetermined sampling rate to generate audio data as high-resolution audio data, and processing of sampling audio outside the predetermined period at the predetermined sampling rate to generate audio data as normal audio data. Also, a reproduction time conversion unit stretches the reproduction time period of the high-resolution audio data.

The subject disclosure is directed towards a technology that may be used in an audio processing environment. Nodes of an audio flow graph are associated with virtual mix buffers. As the flow graph is processed, commands and virtual mix buffer data are provided to audio fixed-function processing blocks. Each virtual mix buffer is mapped to a physical mix buffer, and the associated command is executed with respect to the physical mix buffer. One physical mix buffer mix buffer may be used as an input data buffer for the audio fixed-function processing block, and another physical mix buffer as an output data buffer, for example.

A parallel transfer rate converter inputs first parallel data with number of samples being S1 pieces in synchronism with a first clock, and outputs second parallel data with number of samples being S2=S1(m/p) pieces (p is an integer equal to or larger than 1) in synchronism with a second clock having a frequency which is p/m times of a frequency of the first clock. A convolution operation device inputs the second parallel data in synchronism with the second clock, generates third parallel data with number of samples being S3=S2(n/m) pieces (S3 is an integer equal to or larger than 1) by executing a convolution operation with a coefficient indicating a transmission characteristic to the second parallel data, and outputs the third parallel data in synchronism with the second clock.