Talk:Latency (audio)

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Ideally[edit]

Ideally no longer in article text ~Kvng (talk) 20:06, 5 June 2022 (UTC)[reply]

The following discussion is closed. Please do not modify it. Subsequent comments should be made on the appropriate discussion page. No further edits should be made to this discussion.

The word "ideally" has been brought in via recent edit: "...local circuits should ideally have a latency of 1 millisecond or better." Ideal latency is zero, nonexistant. How else can the recent sentence be stated? Binksternet (talk) 18:38, 30 March 2008 (UTC) OK, but a zero or non-existent latency is not actually achievable in practical digital audio or system designs. Therefore a practical "ideal" statement has to set a range for audio latency such as "1 millisecond or better" which also covers the situation where latencies tend towards zero. While zero audio latency may be the perfect ideal condition, it is not possible to achieve practically so would not be a helpful statement in my view. Theoretically, the only place audio could has zero latency is when measured at the precise point of emission. (i.e. not traveling any distance over any medium.) —Preceding unsigned comment added by Chrisc (talk • contribs) 02:00, 27 January 2009 (UTC)[reply]

The discussion above is closed. Please do not modify it. Subsequent comments should be made on the appropriate discussion page. No further edits should be made to this discussion.

What are the reference points for latency measurements?[edit]

When we talk about latency, we need to condition that by describing the component or system for which the latency is being specified or measured. The lead in this article probably needs to state as much. My recent edits to the lead attempted to enumerate all the potential sources of latency in a system. I included speed of sound as one potential source. User:Binksternet reverted that edit and has stated that this is not a latency contributer. Speed of sound comes into play if you consider propagation through air to be part of your system. With no explicit definition of the system, we have to assume that in some cases it is included and in some cases it isn't. In any case, I would argue that it is always a potential source of latency that should be given consideration. --Kvng (talk) 22:39, 27 June 2010 (UTC)[reply]

Francis Rumsey writes that "latency is the delay incurred in executing audio operations between input and output of a system." I imagine if the system includes air, then the speed of sound is involved. I can see returning speed of sound to the lead. I deleted it earlier because my professional encounters with it are all electronic ones, not natural ones. Binksternet (talk) 05:19, 28 June 2010 (UTC)[reply]
You do need to consider air as part of the audio system and a contributor to latency in cases such as sound reinforcement. --Kvng (talk) 15:30, 28 June 2010 (UTC)[reply]
Certainly sound reinforcement must take speed of sound into consideration, and since that is my career, I am well aware of the physics. It's just that the word latency is not used by practitioners in my field to describe the delay in air, so I have resisted the word's extension that direction. The word latency is instead saved for electronic processes, mainly digital signal processing and format conversion; processes that apply to all listeners. The speed of sound applies differently to each listener depending on position. Again, I grant that the speed of sound is an undeniable part of a system that has air as a sound-carrying medium, so the loosest definition of latency will include the speed of sound. Binksternet (talk) 15:57, 28 June 2010 (UTC)[reply]

1 ms[edit]

I believe it is beneficial to have sub 1 ms latency when monitoring through headphones. The Introduction to Livewire ref does indicate that <3 ms is inaudible. I personally find comb filtering can be an issue starting at 1 ms. 1 ms is, however, difficult to achieve with digital audio systems as AD and DA conversions alone can consume this amount of time. If you can't get below 1 ms it is arguably better to go out to 10 ms than it is to operate in the 1-10 ms area where comb filtering is most annoying. No ref for this yet. --Kvng (talk) 17:49, 21 June 2011 (UTC)[reply]

From empirical research into this I concur, it's very frustrating to have to deal with 5-10 ms latencies when monitoring audio. Even a 1 or 2 ms latency is slightly audible if I pay attention (which I do so rarely... ;-) Chris W. (talk) 22:59, 6 February 2013 (UTC)[reply]
Uhm, you realise that 1ms is the difference in timing you will experience if you move your head about one foot in either direction from the source, right? Low latency *is* important, but mostly because it cascades into higher latency quickly once you've got more than one hop. -- — Preceding unsigned comment added by 121.45.92.47 (talk • contribs)
1 ms is no problem on its own but when mixed with live sound a comb filter is created. A 1 ms comb filter creates an obvious notch at 500 Hz which some headphone wearing DJs describe as feeling like their brains are being sucked out. ~KvnG 14:34, 20 November 2014 (UTC)[reply]
While not quite as crazy as what some people claim they can hear when using oxygen-free USB cables, that still has a very tenuous grip on reality. In order for that to happen, you would need to have _both_ point sources for that 500Hz signal *and* be a very exact distance from *all* of them (including all of their reflections). Since neither musical instruments, nor loudspeakers are point sources (and in most/all cases will radiate sound from a source larger than the wavelength of a 500Hz tone), this seems no more probable than the normal effects that can be seen in any room even without a source delayed by 1ms. Even if you happened to be sitting exactly at the locus of such a local minima for some frequency from one source, your ears are far enough apart that they won't both be in it, and moving your head just a few cm would take you out of it. If you're listening to "live sound" through headphones. then with a 1ms "delay", the sound would actually get to you faster in the headphones than it will from any performer or speaker that is more than one foot away from you. If this effect were as you say, then the performers themselves would also be at risk of having their "brains sucked out" while they played as they moved around :) Sorry, but if you really can hear something, you're going to need a better explanation for it than this. There is nothing magical about 1ms wire latency, phase cancellation depends on the *relative* timing between multiple sources, not the absolute time from any one of them. -- — Preceding unsigned comment added by 14.2.11.249 (talk • contribs)
Point sources and exact level matching are required for a perfect comb filter. We don't need a perfect comb filter to mess up the perceived frequency response in an uncomfortable way. Any motion would not take you out of the comb filter, it just changes the frequency response slightly.
I agree that this effect depends on relative timing. We're talking about a DJ who is, say, 3 inches from the microphone and headphones which are less than an inch from the ears. As the DJ speaks, sound conducts through the bone from the DJs mouth to the ears, that path is perhaps 0.25 ms. The path from mouth to microphone through electronics to headphones to ears is 1.25 ms if we assume 1 ms for the electronics. The difference is 1 ms producing the comb filter with first dip centered at 500 Hz. ~KvnG 14:27, 26 November 2014 (UTC)[reply]
Telos' Livewire protocol (based on multicast RTP/IP) also provides "less than 3 ms"[1][dead link] RTT IP transmission of PCM audio over a LAN and they tout it as the ideal choice for various scenarios (radio, mixing, surround work) to that end. Leo Laporte uses Telos stuff in his studio and seems to be happy with it, and he has a fair deal of background in broadcast and radio. Details: http://axiaaudio.com/livewire
Also this document may be of some relevance (it's interesting reading irrespective): [2][dead link] Chris W. (talk) 22:59, 6 February 2013 (UTC)[reply]
Christopherwoods both the links you provide above are dead (404). ~KvnG 14:37, 20 November 2014 (UTC)[reply]

This discussion goes against the guideline at WP:NOTAFORUM. Let's stick to discussing how to improve the article. Binksternet (talk) 22:05, 23 November 2014 (UTC)[reply]

Sorry, I didn't mean for that explanation to become a side-show. The point in question "local circuits should ideally have a latency of 1 millisecond or better" seems like a relatively uncontroversial statement, though "should ideally have the lowest latency possible" is possibly more strictly accurate. It's never wrong to try to reduce latency, though sometimes there are technical constraints that are in conflict with being able to do so. The idea that there is some black hole between 1ms and 10ms is incorrect though, and should not be included in the article.
I don't find the original statement "dubious", but finding a reliable source to cite for it would indeed be an improvement. -- — Preceding unsigned comment added by 14.2.11.249 (talk • contribs)
I removed the statement under discussion, about "local circuits" which were in any case not defined. Binksternet (talk) 00:40, 24 November 2014 (UTC)[reply]
I think that's a good solution. The previous sentence already covers this, and is less confusing without this uncited qualifier. Thanks. -- — Preceding unsigned comment added by 14.2.2.160 (talk) 20:26, 10 December 2014 (UTC)[reply]

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