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The distinction between FIR and IIR is not recursion. for example, y[n] = y[n-1] + x[n] - x[n-k] is not an IIR filter as it could be rewritten as the sum of the last k samples. The actual distinction is based on the number of non-zero outputs for a non-zero input. ▼
And also, the ideal lowpass "sinc" filter is an IIR filter, though it is commonly written using only inputs samples.▼
<span style="font-size: smaller;" class="autosigned">— Preceding [[Wikipedia:Signatures|unsigned]] comment added by [[Special:Contributions/24.130.9.71|24.130.9.71]] ([[User talk:24.130.9.71|talk]]) 21:20, 9 September 2012 (UTC)</span><!-- Template:Unsigned IP --> <!--Autosigned by SineBot-->▼
:There is a more important difference between y[n] = y[n-1] + x[n] - x[n-k] and y[n] = x[n] + x[n-1] + ... + x[n-k+1].
:With floating-point arithmetic, the recursive implementation has arithmetic error accumulation (random-walk) as n increases. The FIR implementation does not.
:--[[User:Bob K|Bob K]] ([[User talk:Bob K|talk]]) 13:47, 18 September 2016 (UTC)
== Intro reads like an app note, not a text book ==
{{Archive top |result=No longer in article ~[[User:Kvng|Kvng]] ([[User talk:Kvng|talk]]) 16:33, 3 April 2020 (UTC)}}
:The goal of DSP is usually to measure, filter and/or compress continuous real-world analog signals. '''The first step is usually to convert the signal from an analog to a digital form, by sampling and then digitizing it using an analog-to-digital converter (ADC), which turns the analog signal into a stream of numbers. However, often, the required output signal is another analog output signal, which requires a digital-to-analog converter (DAC).''' Even if this process is more complex than analog processing and has a discrete value range, the application of computational power to digital signal processing allows for many advantages over analog processing in many applications, such as error detection and correction in transmission as well as data compression.
I think this paragraph shows evidence of making sense, but doesn't. This should stick to first princples (sampling and quantization), or discuss what could be described as "degrees of freedom over domains of operation" (the group of all applications).
As it is, it's been hacked to one odd class of applications. On a music CDROM it's called "ADA". My analog outputs are usually PWM. [[User:JohnPritchard|JohnPritchard]] ([[User talk:JohnPritchard|talk]]) 16:16, 2 January 2013 (UTC)
{{Archive bottom}}
==
▲The distinction between FIR and IIR is not recursion. for example, y[n] = y[n-1] + x[n] - x[n-k] is not an IIR filter as it could be rewritten as the sum of the last k samples. The actual distinction is based on the number of non-zero outputs for a non-zero input.
This article has languished for too long. Complaining about things you dislike will not make it better. Start writing. I will begin on the lead - which sucks btw. [[User:Codwiki|Codwiki]] ([[User talk:Codwiki|talk]]) 16:19, 7 July 2015 (UTC)
▲And also, the ideal lowpass "sinc" filter is an IIR filter, though it is commonly written using only inputs samples.
== Time and space domains ==
▲<span style="font-size: smaller;" class="autosigned">— Preceding [[Wikipedia:Signatures|unsigned]] comment added by [[Special:Contributions/24.130.9.71|24.130.9.71]] ([[User talk:24.130.9.71|talk]]) 21:20, 9 September 2012 (UTC)</span><!-- Template:Unsigned IP --> <!--Autosigned by SineBot-->
It seems as though most of the content of [[Digital_signal_processing#Time_and_space_domains|this section]] is about [[Digital filter]], not [[Time ___domain]]. I think I could find a place for the material in [[Digital filter]] and then summarize [[Time ___domain]] in this section and also add something about space ___domain for digital imaging processing. I'm happy to do this [[WP:BOLD]]ly; not looking for permission. Mostly posting here as a reminder to myself. But, if anyone has any objections or suggestions, feel free to respond. ~[[User:Kvng|Kvng]] ([[User talk:Kvng|talk]]) 16:30, 13 February 2018 (UTC)
{{done}} ~[[User:Kvng|Kvng]] ([[User talk:Kvng|talk]]) 20:39, 24 August 2020 (UTC)
== FIR filters have many advantages, but are computationally more demanding ==
{{u|M.perkins}} changed FIR to IIR and I had to look carefully to recognise this was wrong. I removed the whole sentence because the section is about Z-plane analysis, not computational efficiency and the sentence does not explain the advantages of an FIR filter. {{u|Dicklyon}} reverted to the original. I beleive these facts are already covered in [[Finite impulse response]] and [[Infinite impulse response]] and teasing them in this section is unnecessary and appears to be causing some confusion. ~[[User:Kvng|Kvng]] ([[User talk:Kvng|talk]]) 15:45, 26 March 2019 (UTC)
:OK with me if you take it out after considering the correction I reverted to. [[User:Dicklyon|Dicklyon]] ([[User talk:Dicklyon|talk]]) 02:42, 27 March 2019 (UTC)
::[https://en.wikipedia.org/w/index.php?title=Digital_signal_processing&type=revision&diff=890154422&oldid=889500724 Removed] ~[[User:Kvng|Kvng]] ([[User talk:Kvng|talk]]) 14:08, 30 March 2019 (UTC)
== Uncertainity ==
Sorry, Heisenberg wasn’t just talking about particles.. https://pages.hmc.edu/ruye/e161/lectures/fourier/node2.html#:~:text=in%20frequency%20domain%20provides%20complete,the%20signal%2C%20but%20never%20both. [[User:Greglocock|Greglocock]] ([[User talk:Greglocock|talk]]) 20:07, 26 July 2025 (UTC)
:Thanks, I did not know this. ~[[User:Kvng|Kvng]] ([[User talk:Kvng|talk]]) 23:15, 26 July 2025 (UTC)
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