Software effect processor: Difference between revisions

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{{Short description|Computer program for audio processing}}

The '''software effect processor''' is a computer program which is able to modify the sound coming from a source of sound in real-time.

{{More citations needed|date=January 2024}}[[Image:Cubase 6 feature - software instruments and software effects.svg|thumb|250px|top: [[Software synthesizer|Software instruments]], and <br />bottom: Software effect processors<br />on [[Steinberg Cubase|Cubase]] 6]]

A '''software effect processor''' is a computer program that alters the sound from a [[Digital audio|digital]] source through [[audio signal processing]] in [[Real-time computing|real time]]. It is a digital analog of hardware [[Effects unit|effects processors]]. It is an integral part of [[audio editing software]], such as in [[Adobe Audition]]<ref>{{Cite book |last=Reese |first=David |title=Audio Production Worktext: Concepts, Techniques, and Equipment |last2=Gross |first2=Lynne |last3=Gross |first3=Brian |date=Nov 12, 2012 |publisher=Taylor & Francis |isbn=9781136035531 |at=Chapter 8.3}}</ref>

==Principle of operation==

The digital audio signal, fromwhose theorigin inputmay isbe transformedanalog (by conversion to thedigital) or be in an already digital source (asuch streamas ofan numbers)audio infile, audioor hardwarea and[[software passedsynthesizer]]), tois astored in temporary pieceallotments of computer memory, called [[Data buffer|buffers]]. ThenOnce itthere, isthe modifiedsoftware effect processor modifies the signal according to a specific algorithm, which creates the desired effect. After this operation, the signal ismay be transformed from digital to analog and sent to thean audible output, stored in digital form for later reproduction or editing, or sent to other software effect processors for additional processing.

#The larger the buffer is, the more time it takes to play the audio data sent for playback. Large buffers increase the time required before the next buffer can be played, this delay is usually called latency. Every system has certain limitations - too small buffers involving negligible latencies cannot be smoothly processed by computer, so the reasonable size starts at about 32 samples. The processor load does not affect latency directly (it means, once you set certain buffer size, the latency is constant),. butBut with very high processor loads, the processingbuffer startsisn't filled with new sound in time for playback, and the sound droppingdrops out. Increasing buffer size or quitting other applicationapplications helps to keep playback smooth.

== Support in operating systems ==

The default Windows drivers are not optimized for low latency effect processing. As a solution, [[Audio Stream Input/Output]] (ASIO) was created. ASIO is supported by most professional music applications. Most sound cards directed at this market support ASIO. If the hardware manufacturer doesn't provide ASIO drivers, there are other [[Audio_Stream_Input/Output#Free_alternatives|ASIO free alternatives]], which can be used for any audio interface. ASIO drivers can be emulated, in this case the driver name is ASIO Multimedia. However, the latency when using these drivers is very high.

All the Mac compatible hardware uses CoreAudio drivers, so the software effects processors can work with small latency and good performance.

== See also ==

* [[Audio plug-in]]

== References ==

{{Reflist}}

==External links==

* [https://global.oup.com/us/companion.websites/fdscontent/uscompanion/us/static/companion.websites/9780199922963/Chapter8.html Software effect processor demo in companion with ''Refining Sound: A Practical Guide to Synthesis and Synthesizers'']

{{Music technology}}

[[Category:Music software]]

===Latency===

[[Category:Pitch modification software]]

'''Latency''' is the main issue of real-time audio processing. It represents delay between reception of the input signal and its transmitting to the output. The larger the buffer is, the more time it takes to fill it by digital audio data. Large buffers increase the time required for processing audio in computer, this delay is usually called latency. You can adjust the buffer size and set the latency in Preferences of inTone or other applications.

# Every system has certain limitations - too small buffers involving negligible latencies cannot be smoothly processed by computer, so the reasonable size starts at about 32 samples. The processor load does not affect latency directly (it means, once you set certain buffer size, the latency is constant), but with very high processor loads the processing starts dropping out. Increasing buffer size or quitting other application helps to keep playback smooth.