Talk:Latency (audio)

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I read somewhere that 13ms is an in audable latency amount to trained ears. Wonder if someone knows of any articles relating to this. —Preceding unsigned comment added by 207.54.108.113 (talk) 19:32, August 24, 2007 (UTC)

I did some measurements of professional live production systems in the BBC some years ago (Ref: page 14 of http://www.bbc.co.uk/rd/pubs/whp/whp-pdf-files/WHP074.pdf). I found that within a production centre about 1.5 milliseconds each way was common and in wide area around 9 to 10 milliseconds each way was acceptable between different production centres provided that audio jitter (or latency variation) performance is kept below 5 microseconds for production audio paths and within 125 microseconds more generally. Keeping the jitter low is vital in order to keep the audio data receive buffering low and thus minimise end to end latency. All this work was done to provide the specifications for the development of linear wide area low latency audio standards and this led to the development of AES47. This standard has now been installed across the BBC to provide high quality audio contributions between all major broadcast centres across the UK and does in practice provide latencies of around 9 milliseconds. —Preceding unsigned comment added by Chrisc (talk • contribs) 21:49, 24 March 2008 (UTC)

The article appears to be biased against Windows. —Preceding unsigned comment added by 77.86.113.168 (talk) 08:48, 6 February 2008 (UTC)

This is not biased against Windows. It merely points out that windows does have more issues with latency, and describes the usual solution. Latency is the main problem for Windows audio. For Linux, it's software and plugins... Ab8uu (talk) 15:01, 26 February 2008 (UTC)

I edited the page to remove a confusing ambiguity. "keyboard connected to a computer" brings to mind the thing I used to write this comment. Looking at the link, it is "musical keyboard", so I changed the link to reflect that. Ab8uu (talk) 15:01, 26 February 2008 (UTC)

[edit] Recent edits regarding live event audio

The new section "Audio latency in live performance venues" contains some problematic assumptions. I don't think the speed of sound in air has much to do with the topic of latency. There's no great problem associated with concert sound in an arena and digital latency, unless the venue is very small and the latency is very large, as it might be in a speaker management system that uses digital FIR filtering for each of the pass bands including the lowest frequencies where the latency is largest. At the most extreme, a latency of 30ms would be fairly easy to accommodate on an expansive arena stage; one would simply time the main speakers back to the kick drum or bass guitar ten (or so) meters away. The speed of sound in air would be equalled by such a long latency. The more common situation of about two to ten milliseconds of IIR filter latency is very easily dealt with at concert venues of all sizes. Perhaps this section would read better if greatly reduced in size, eliminating every mention of speed of sound, humidity in air, absorption of high frequencies in air and venue reverberation. Note that audio cables of all sorts have insignificant latency. Binksternet (talk) 22:18, 24 March 2008 (UTC)

I do agree that the speed of sound in air has a marginal connection with audio latency in systems generally speaking. I was hoping to try and link the design needs of digital audio systems such as those used in live venues to tangible benchmarks and all the multi-speaker sound reinforcement designs I have been involved with over the years do indeed make this correlation. (comb filter effect) In rigs where additional PA stacks are placed half-way down the auditorium in order to keep the front stack SPL down to safe limits. (where they could not be “sky hooked”) These additional stacks would be timed at around 3 mS per metre from the stage and would have appropriate latency inserted into the amps feeding these. I think this is a good example of designed in latency and its link to the speed of sound in air. Currently OFCOM in the UK is trying to force the entertainment industry here to move to using digital radio microphone and in-ear monitoring and failing due to the latency and cost of such a move is a good example of why latency management is vital in PA structures. I was just trying to be complete in mentioning absorption, reverberation and so on and agree all that could come out if you feel it should. Yes of course audio cables have negligible latency however a number of modern digital replacements do have a small impact. Maybe this should be reworded to convey this in a better way? As always with digital structures, it not that any one section is necessarily an issue, but the accumulation end to end that produces the problems. This is a great subject and I would like to help get it as useful as possible. If you were the person who started this up, well done! —Preceding unsigned comment added by Chrisc (talk • contribs) 23:24, 24 March 2008 (UTC)

Speaker systems that are delayed on purpose don't fall under the topic of latency, in my opinion. Latency is the unwanted by-product of digital sound processing. Anything done on purpose is just sound system design. The part of latency that intersects with sound system design is the part that interferes with hoped-for goals or that forces the designer away from one sound system architecture and toward another.

Even digital audio cables don't have significant latency, though the sending and receiving circuits on either side do. I've measured ADAT latency at 96kHz sampling rate to be about 0.5 ms on one particular product.

Digital radio latency does deserve a mention here, as does normal wired/wireless digital processing latency that makes the option of in-ear monitors less attractive to some users. Whatever you have on OFCOM's lack of success in persuading the entertainment industry would be helpful here.

One thing that's not mentioned here (so far) is the latency that is associated with each processing block within a DSP. When I assign a new EQ filter, compressor or limiter element to a generic DSP unit (such as BSS Soundweb, A&H iDR, Peavey MediaMatrix etc.), I get a slightly longer total latency. This topic definitely deserves more effort from all of us. I'll be able to dig in within a few days but right now, no. Feel free to sculpt it as you see fit. Binksternet (talk) 23:54, 24 March 2008 (UTC)

I see where you are coming from. However, while sound system design latency is expected, it provides the constraints by which equipment has to work and system designers have to be very aware of these. Sound system design is also an important example of where audio latency is a key factor. You mention a very important point with the issue of the changing latency with the insertion of additional block of processing. I am sure there is a good bit more to be explored here. I will do some digging around on the digital radio issues and see what I can come up with over the next week or two. —Preceding unsigned comment added by Chrisc (talk • contribs) 00:32, 25 March 2008 (UTC)

I've been thinking about this section quite a bit and it just seems to be overwhelmingly dire in tone. Digital audio latency is not that big a deal in live performance spaces. It's noted, then incorporated into system design. After that it isn't a concern.

Though I have very few references, I'm rewriting the performance section (which at this time has NO references.) This version is rushed and could use more additional information. I'm going to leave all the straight delay parts out--any discussion of delay adjustments made on purpose to live performance loudspeakers should go over to Delay (audio effect)#Straight delay, or to its own article. Binksternet (talk) 20:43, 30 March 2008 (UTC)

[edit] Ideally

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)