摘要:
A device, comprising a first interpolator that is configured to (a) receive, at a first clock rate, a first signal having a first sampling rate and (b) output, at a second clock rate, a second signal having a first desired sampling rate average; wherein the first interpolator comprises: a first buffer for storing the first signal; and a first fractional sampling ratio circuit that is configured to generate a first pattern of fixed point values, wherein an average value of the first pattern corresponds to a first desired sampling rate ratio between the first desired sampling rate average and the first sampling rate.
摘要:
Embodiments of apparatuses, articles, methods, and systems for efficient generation of a time domain signal in multi-carrier communications are generally described herein. Other embodiments may be described and claimed.
摘要:
The subject disclosure is directed towards dynamically computing anti-aliasing filter coefficients for sample rate conversion in digital audio. In one aspect, for each input-to-output sampling rate ratio (pitch) obtained, anti-aliasing filter coefficients are interpolated based upon the pitch (e.g., using the fractional part of the ratio) from two filters (coefficient sets) selected based upon the pitch (e.g., using the integer part of the ratio). The interpolation provides for fine-grained cutoff frequencies, and by re-computation for each pitch, smooth anti-aliasing with dynamically changing ratios.
摘要:
Embodiments of apparatuses, articles, methods, and systems for efficient generation of a time domain signal in multi-carrier communications are generally described herein. Other embodiments may be described and claimed.
摘要:
Data rate conversion devices and methods are provided. A method for converting a first digital signal having a first sampling rate into a second digital signal having a sampling rate close to a predetermined second sampling rate comprises the following operations: when the ratio of the first sampling rate to the second sampling rate is a repeating infinite decimal, calculate at least two calibrating coefficient values and output the calibrating coefficient values according to a predetermined rule; conduct overflow operation on the output calibrating coefficient; and interpolate the first digital signal using the output calibrating coefficient and the result of the overflow operation to obtain the second digital signal such that during any period of a certain length along time axis, sampling times of the second digital signal equals to sampling times of the second sampling rate.
摘要:
A cascaded integrator comb filter includes a first integrator that receives an input signal x[n] and provides an integrated signal, and a fractional integrator that also receives the input signal x[n] and provides a fractional integrated signal. A summer sums the integrated signal and the fractional integrated signal and provides a summed signal indicative thereof to a second integrator, which receives and integrates the summed signal to provide a second integrator output signal. A decimator unit receives the second integrator output signal and provides a decimated signal to a differentiator that receives the decimated signal and provides a differentiated signal.
摘要:
A circuit for single or parallel digital fractional interpolation of data samples has a fractional interpolator filter, an oscillator for outputting timing signals to the fractional interpolator filter, and a detector loop with a strobe feedback from the oscillator for outputting a frequency adjustment to the oscillator. Three different approaches are shown to determine the frequency adjustment. One approach is to generate a pulse based on the symbol clock, and measure the differences between the pulse and the strobe and between the strobe and the pulse. The smaller is the frequency adjustment. Another approach is to adjust the strobe period to match the symbol clock period. A third approach is to add an oscillator-driven clock to the symbol clock and integrate the sum over a symbol clock period to generate the frequency adjustment. Preferably, the interpolator filter takes N parallel inputs and samples each in parallel based on a plurality of oscillator timing signals, each corrected with reference to the frequency adjustment.
摘要:
Sample rate converters (12) for converting input sample rates (F81) of signals into output sample rates (Fs4) are provided with sample rate adapters (3,6) for adapting (basic idea) intermediate sample rates (Fs2) such that output sample rates (Fs4) are larger (upsampling) or smaller (downsampling) than input sample rates (F81), to reduce their complexity and to avoid bookkeeping and structure switching problems. Sample rate adapters (3,6) in the form of variable sample rate decreasers (3) allow the sample rate converters (12) to be used in video applications requiring DC-out being equal to DC-in. Sample rate adapters (3,6) in the form of variable sample rate increasers (6) allow the sample rate converters (12) to be used in audio applications. By locating the sample rate adapter (3,6) between a fixed sample rate increaser (1) for increasing with a factor K and a fixed sample rate decreaser (5) for decreasing with a factor M, filters (2,4) in between can be designed independently from the varying factor L as long as K and M are fixed and L
摘要翻译:将采样速率转换器(12)用于将信号的输入采样率(F 81)变换成输出采样率(F S s S S S S S S N) 用于适应(基本思想)中间采样率(F S2),使得输出采样率(F S S S S S S)比输入更大(上采样)或更小(下采样) 采样率(F 81),以降低它们的复杂性并避免记帐和结构切换问题。 可变采样率降压器(3)形式的采样率适配器(3,6)允许采样率转换器(12)用于需要DC-out等于DC-in的视频应用中。 可变采样率增加器(6)形式的采样率适配器(3,6)允许采样率转换器(12)用于音频应用。 通过将采样率适配器(3,6)定位在用因子K增加的固定采样率增加器(1)和以因子M减小的固定采样率递减器(5)之间,滤波器(2,4)在 只要K和M是固定的,可以独立于变化因子L设计,并且 / SUP> -K。
摘要:
A method and a computer program product for sample rate conversion that features distributive or hybrid filtering to reduce unwanted artifacts, such as aliasing and the computational requirements to avoid the aforementioned artifacts. The method includes receiving, at a first sample rate, a plurality of data points, associated with a first signal, operating on the plurality of data points to associate the signal with a predetermined set of parameters, with the set of parameters including a first transition band having an image associated therewith; and varying the sample rate associated with the first signal by interpolation with an interpolator having associated therewith a second transition band, with the width associated with the second transition band being a function of a spectral separation between the first transition band and its image, wherein a second signal is produced having a sequence of data samples approximating the first signal.
摘要:
A sample rate converter reduces the sampling rate of a signal by a fractional number U/D, where U represents an up-sampling rate and D represents a down-sampling rate. The converter comprises an input for receiving an input data stream at a first rate and an FIR filtering stage. The FIR filtering stage comprises a set of D polyphase filter branches, each branch including a set of filter coefficients which operate on a sample of the input signal. The converter also comprises a commutative switch which selectively connects a sample of the input data stream to one of the polyphase filter branches, the switch being arranged to skip every U−1 filter branches during a cycle through the filter branches. An output outputs an output data stream at a second data rate which is lower than the first data rate.