2 edition of Sample rate reduction by decimation found in the catalog.
Sample rate reduction by decimation
James R Yoder
by Dept. of Energy, Sandia Laboratories, for sale by the National Technical Information Service] in Albuquerque, N.M, [Springfield, Va
Written in English
|Statement||James R. Yoder, Data Planning Division 2425, Sandia Laboratories ; prepared by Sandia Laboratories for the United States Department of Energy|
|Series||SAND ; 79-0357|
|Contributions||United States. Dept. of Energy, Sandia Laboratories, Sandia Laboratories. Data Planning Division 2425|
|The Physical Object|
|Pagination||29 p. :|
|Number of Pages||29|
Polyphase Filters Polyphase is a way of doing sampling-rate conversion that leads to very efficient implementations. But more than that, it leads to very general viewpoints that are useful in building filter banks. Before we delve into the math we can see a lot just by looking at the structure of the filtering–. Prolonged transposed polynomial-based filters for decimation Abstract: If sample rate conversion (SRC) is performed between arbitrary sample rates, then the SRC factor can be a ratio of two very large integers or even an irrational number.
When you increase sampling rate, you spread this total (in freq domain) noise energy over a larger bandwidth relative to the sampled signal. Thus, in-band SNR increases. And the rate of increase is 3dB per doubling of sampling rate. You can improve this scaling by doing Sigma-Delta, etc., explained in the first article I linked. I love this topic:D. Performance Analysis of Fractional Sample Rate Converter Using Audio Applications DOI: / | Page Figure3: conversion of discrete sequence x (n) to another discrete sequence y(k) II (ii)Decimation.
I am looking at other avenues to reduce the sampling rate or increase the sampling period and a search in the litterature has yielded decimation as a possible avenue. A quick hack in matlab using the decimate function yields the below figures for a decimated signal after being sampled at Hz. Video Lecture on Step for Sampling Rate Conversion Method in Multi Rate Signal Proccessing from Multirate Signal Processing chapter of Discrete Time .
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Get this from a library. Sample rate reduction by decimation: procedures and effects. [James R Yoder; United States. Department of Energy.; Sandia Laboratories.; Sandia Laboratories. Data. Currently, decimation is the term used for reducing the sample rate by any integer factor.) When the sampling rate is being increased, the process is known as interpolation, i.e., estimating intermediate sample values.
Because decimation is the simplest of the two rate. Decimation reduces the original sample rate of a sequence to a lower rate. It is the opposite of interpolation.
decimate lowpass filters the input to guard against aliasing and downsamples the result. The function uses decimation algorithms and from.
> >A good book for reference on this topic is > >Multirate Systems and Filterbanks by P.P. Vaidyanathan. > > > >Cheers > >Bhaskar > > Hi Bhasker, > yep, below are a few other references for seb. > I'll bet he doesn't realize that a person could > make a career out of learning all the theory of > sample rate changing.
Loosely speaking, “decimation” is the process of reducing the sampling rate. In practice, this usually implies lowpass-filtering a signal, then throwing away some of its samples.
“Downsampling” is a more specific term which refers to just the process of throwing. DECIMATION A reduction in the sampling rate by factor M is achieved by discarding every M-1 samples or equivalently keeping every Mth sample.
While discarding M-1 of every M input samples reduces the original sample rate by a factor of M, it also causes input frequencies above one-half the decimated sample rate to be aliased into the fre. Bachelor Thesis Sample Rate Conversion in Digital Signal Processors conducted at the Signal Processing and Speech Communications Laboratory Graz University of Technology, Austria by Marian Forster, Supervisor: DI Dr.
Werner Magnes Graz, system are decreasing (decimation) and increasing (interpolation) the sampling-rate of a signal. Multirate systems are sometimes used for sampling-rate conversion, which involves both decimation and interpolation.
Decimation Decimation can be regarded as the discrete-time counterpart of sampling. Whereas in sampling we start with a.
The problem is there really is no guarantee that your received signal is 40MHz, esp. in communication where additive white noise (AWGN) is a big part of the model. If you sampled at 80MHz, but AWGN is significant at higher frequencies than 40MHz. In single-rate DSP systems, all data is sampled at the same rate no change of rate within the system.
In multirate DSP systems, sample rates are changed (or are different) within the system Multirate can offer several advantages reduced computational complexity reduced transmission data rate.
Digital Signal Processing – p.3/ For example, a sample rate increase by a factor of can be performed by an interpolation of M = 57 followed by a decimation of D = 8, because = 57/8. This M/D sample rate change is illustrated as the processes shown in Figure (a).
The upsampling operation M means insert M – 1 zero-valued samples between each xold(n) sample. The. Using a Sample and Hold (Polivoks Modulator) as a bitcrusher/sample rate reducer. I'm wondering what the practical differences are between sampling rates and decimation. Of course sampling rates will affect the bandwidth required between the device and the software, but following the discussions on rather low sample rates what might be the practical (or the coding difference) between, say 48 kHz sampling rate, and kHz sampling with a x4 decimation.
Two major design challenges of finding the proper number of stages as well as the appropriate sampling rate reduction per stage of decimation filter have so far been the topic of discussions. Description. The Downsample block decreases the sampling rate of the input by deleting samples.
When the block performs frame-based processing, it resamples the data in each column of the M i-by-N input matrix independently. When the block performs sample-based processing, it treats each element of the input as a separate channel and resamples each channel of the input array across time.
The simplest interpolation filter is the zero-order hold (ZOH), which instead of zero stuffing the low-rate input data to produce high-rate data simply holds each sample of the low-rate data for N. On top of the picture we see spectrum of source audio signal with sampling rate Fd.
Spectrum on probation contains 2 areas (blue and light blue). We see what light blue placed above Fd/4. We will decrease (divide 2 times) sample rate source audio and watch to metamorphoses of its spectrum. New sample rate is Fd new = Fd / 2. Possible avoid it. Yes. Decimation is a way to reduce the number of samples delivered to the software.
I believe that the minimum sample rate of the hardware is 2M samples per second witch is huge for a software designed to use a top grade sound card limited to KHz sampling but with at least 16 bits resolution. $\begingroup$ Indeed "then later the sample rate is increased somehow" is quite vague. A whole data processing chain can be large.
A whole data processing chain can be large. If a decimation at link 1, followed by some upsamling at link 4, is globally beneficial, then it is worth the use.
I purchased the book "Multirate Signal Processing for Communication Systems" by Fredric J. Harris because it contained an entire chapter devoted to the Cascaded Integrator Comb (CIC) filter. This filter, sometimes referred to as the Hogenauer Filter, can be used to implement efficient multirate decimation and interpolation filters for large Reviews:.
Decimation is a method of observing only a periodic portion of the ADC samples, while ignoring the rest. The result is to reduce the sample rate of the ADC. For example, a decimate-by-4 mode would mean (total samples)/4, while all other samples are effectively discarded.Both the interpolation and decimation filters incorporate a low-pass filtering function.
The reason for this LPF, however, is quite different for each case. For decimation, the LPF serves to eliminate high frequency components in the spectrum.
If these components were not filtered out, they would alias when the reduction in sample rate is.Sample rate reduction is realized by transforming the transfer function and invoking the noble identity of multirate signal processing.
In this technique, an Nth-order recursive equation is decomposed into first- and second-order parallel sections. Computational efficiency is enhanced due to parallel implementation.