摘要:
A decimation system comprising a plurality, L, of cascaded Finite Impulse Response (FIR) decimation filters. Each decimation filter has a transfer function of the form H(z)null(1nullznull1)N, where N is an integer. Each FIR decimation filter performs decimation by a common factor I. The cascaded FIR decimation filters together achieve a decimation result substantially identical to that of an Nth-order CIC filter (that is, a CIC filter having N integrator stages) that performs decimation by a factor IL.
摘要翻译:一种抽取系统,包括多个级联的有限脉冲响应(FIR)抽取滤波器。 每个抽取滤波器具有H(z)=(1 + z -1)N形式的传递函数,其中N是整数。 每个FIR抽取滤波器通过公共因子I执行抽取。级联FIR抽取滤波器一起实现与执行N次级CIC滤波器(即,具有N个积分器级的CIC滤波器)基本相同的抽取结果,其执行 抽取因子I 。
摘要:
A simplified algorithm for digital signal interpolation and a novel architecture to implement the algorithm in an integrated circuit (“IC”) with significant space constraints are presented. According to embodiments of the present invention, the interpolator is divided into two parts. The first part of the interpolator increases the sample rate by a factor of two and smoothes the signal using a half-band Infinite Impulse Response (“IIR”) filter. The second part of the interpolator increases the sample rate of the signal by a factor of thirty-two using a zero-order-hold (“ZOH”) circuit. In one embodiment, the half-band IIR filter is implemented using an all-pass lattice structure to minimize quantization effects. The lattice coefficients are chosen such that the structure can achieve all filter design requirements, yet is capable of being implemented with a small number of shifters and adders, and no multipliers.
摘要:
A decimation system comprising a plurality, L, of cascaded Finite Impulse Response (FIR) decimation filters. Each decimation filter has a transfer function of the form H(z)=(1+z−1)N, where N is an integer. Each FIR decimation filter performs decimation by a common factor I. The cascaded FIR decimation filters together achieve a decimation result substantially identical to that of an Nth-order CIC filter (that is, a CIC filter having N integrator stages) that performs decimation by a factor IL.
摘要翻译:一种抽取系统,包括多个级联的有限脉冲响应(FIR)抽取滤波器。 每个抽取滤波器具有形式为H(z)=(1 + z 0 - 1)的传递函数,其中N是整数。 每个FIR抽取滤波器通过公共因子I执行抽取。级联FIR抽取滤波器一起实现与第N个次级CIC滤波器(即,具有N个N个CIC滤波器的CIC滤波器)基本相同的抽取结果 积分器级),其通过因子I L执行抽取。
摘要:
A method is provided for decimating a digital signal by a factor of M and matching it to a desired channel bandwidth. The method applies the digital signal input samples to a (M−1) stage tapped delay line, downsamples the input samples and the output samples of each tapped delay line stage by a factor of M, and applies each of the M downsampled sample value streams to M allpass IIR filters, respectively. The M allpass IIR filtered sample streams are then summed and scaled by a factor of 1/M. The result can then be filtered by a digital channel filter.
摘要:
A method is provided for decimating a digital signal by a factor of M and matching it to a desired channel bandwidth. The method applies the digital signal input samples to a (M-1) stage tapped delay line, downsamples the input samples and the output samples of each tapped delay line stage by a factor of M, and applies each of the M downsampled sample value streams to M allpass IIR filters, respectively. The M allpass IIR filtered sample streams are then summed and scaled by a factor of 1/M. The result can then be filtered by a digital channel filter.
摘要:
A simplified algorithm for digital signal interpolation and a novel architecture to implement the algorithm in an integrated circuit (“IC”) with significant space constraints are presented. According to embodiments of the present invention, the interpolator is divided into two parts. The first part of the interpolator increases the sample rate by a factor of two and smoothes the signal using a half-band Infinite Impulse Response (“IIR”) filter. The second part of the interpolator increases the sample rate of the signal by a factor of thirty-two using a zero-order-hold (“ZOH”) circuit. In one embodiment, the half-band IIR filter is implemented using an all-pass lattice structure to minimize quantization effects. The lattice coefficients are chosen such that the structure can achieve all filter design requirements, yet is capable of being implemented with a small number of shifters and adders, and no multipliers.
摘要:
A filter structure including a notch filter is designed to have a transmission zero at a frequency slightly displaced from one half of the lower sampling rate of the interpolator/decimator, and thus providing increase in attenuation at the half-rate frequency. The notch filter is comprised of two like all-pass-network filters and has feed forward and feedback connections, the latter connection including a coefficient multiplier. The feedback connection is made between a tapped output of the second of the network filters and an input node. The tapped filter is characterized by throughput and a tapped output transform functions X(Z) and Y(Z) given by the following expressions:X(Z)=[Z.sup.-1 -K]/[1-KZ.sup.-1 ; and,Y(Z)=.alpha.Z.sup.-1 /[1-KZ.sup.-1 ;where Z.sup.-1 is the unit delay operator, K the multiplier coefficient and, .alpha., a structure dependant constant.
摘要:
A method is provided for decimating a digital signal by a factor of M and matching it to a desired channel bandwidth. The method applies the digital signal input samples to a (M−1) stage tapped delay line, downsamples the input samples and the output samples of each tapped delay line stage by a factor of M, and applies each of the M downsampled sample value streams to M allpass IIR filters, respectively. The M allpass IIR filtered sample streams are then summed and scaled by a factor of 1/M. The result can then be filtered by a digital channel filter.
摘要:
Methods and systems for processing multipath, multi-rate digital audio signals are disclosed herein. Aspects of the method may comprise up-sampling a number of digital audio signals and converting the up-sampled digital audio signals to a common data rate prior to mixing. The digital audio signals sampling rates may be converted utilizing half-band interpolators. The up-sampling may also utilize infinite impulse response interpolators. A portion of the up-sampled digital audio signals may have a common data rate. The common data rate may be determined by the number half-band interpolators utilized. Distortion in the digital audio signals may be compensated utilizing an infinite impulse response filter or a finite impulse response filter.
摘要:
A method is provided for decimating a digital signal by a factor of M and matching it to a desired channel bandwidth. The method applies the digital signal input samples to a (M−1) stage tapped delay line, downsamples the input samples and the output samples of each tapped delay line stage by a factor of M, and applies each of the M downsampled sample value streams to M allpass IRR filters, respectively. The M allpass IRR filtered sample streams are then summed and scaled by a factor of 1/M. The result can then be filtered by a digital channel filter.