How Fast Does the Computer Needs to Be for No Latency Issues?

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ftxn

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The Avid (for Pro Tools) qualified computers are expensive and highly powerful computers. I have been looking for something inexpensive. Though, I am looking for something which lets me have no synchronization issues when recording and during playback.

Maybe some people who have had experience with this could let me know. Would an Intel Quad Core be enough when it comes to this? Would I have no latency issues with an Intel Quad Core? Perhaps with Reason and with Pro Tools?

Thanks for the suggestions.
 
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The speed of your CPU is only part of the equation - and "quad core" is a very broad definition, there are probably about 50 different quad core processors in Intel's lineup alone (but frankly, good latency figures should be attainable with just about any modern CPU given that the system isn't constantly under a lot of strain). The biggest latency performance improvements are gonna be found in a good audio interface with drivers to match.
 
Thanks for the answer krushing!

I have tried the recent version of Reason with an Intel Pentium Dual core and I have the Yamaha Audiogram 3 audio interface with its own ASIO drivers and I have around 20 ms input and output latency. Would you say it can be that way? Making music with this is barely possible.

I was looking for something where I can be more sure that I would be getting no latency issues. Though I have been looking for something inexpensive. Whereas I mentioned those Avid qualified computers, which are highly powerful and expensive.
 
With the post I wrote before. Why would the Yamaha Audiogram 3 not be doing it? I am getting around 20 ms input and output latency and that with an Intel Pentium Dual Core.
 
20ms combined or 20ms on both input and output? Judging from a quick googling, it seems that the Audiogram 3's latency performance isn't exactly stellar. Have you adjusted the buffer size?
 
I am having a latency of 20 ms on both (each) with 512 samples. If I adjust the buffer size till down to 64 samples then I am seeing 7ms there, yet when recording and during playback it is not really much better than it was with 20 ms (and 512 samples).

Could the audio interface really be the issue? I was set on the thought that I may definitely need a faster computer.
 
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Hmm, 14ms total latency should be pretty ok, actually - as in "barely noticeable". Are you sure it's actually working as per those figures? What kind of issues are you having, exactly?
 
krushing said:
Are you sure it's actually working as per those figures? What kind of issues are you having, exactly?
Click to expand...

Thanks for the help, the help really does help. Just now I adjusted it to 256 samples and I am having around 12 ms latency on each input/output. I am using my ears to check for this and the issue I am having is that I am hearing synchronization issues during playback. It gets a bit difficult to check for the recording latency, yet I will think of your tip that 14 ms may be pretty ok for this. Though, I also use the piano roll to check for this and when I for example record a piano then the notes do align on the grid in the piano roll, though when I listen to the beat then the piano is not synchronous with the drum beat, the piano is a bit off to the drum beat.

Basically, when I am listening to the playback the notes of the instruments are not synchronous to each other, which means that the notes come a bit too early or too late in the composition.

What would you say where the problem may be with this?
 
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ftxn said:
Thanks for the help, the help really does help. Just now I adjusted it to 256 samples and I am having around 12 ms latency on each input/output. I am using my ears to check for this and the issue I am having is that I am hearing synchronization issues during playback. It gets a bit difficult to check for the recording latency, yet I will think of your tip that 14 ms may be pretty ok for this. Though, I also use the piano roll to check for this and when I for example record a piano then the notes do align on the grid in the piano roll, though when I listen to the beat then the piano is not synchronous with the drum beat, the piano is a bit off to the drum beat.

Basically, when I am listening to the playback the notes of the instruments are not synchronous to each other, which means that the notes come a bit too early or too late in the composition.

What would you say where the problem may be with this?
Click to expand...

Truthfully, it's hard for the average person to tell the difference between 12ms and 20ms of latency.

Sounds like this not a latency issue at all. Inspect the notes in the piano roll, you might be surprised to see that they're not dead on, they might be slightly off but it wouldn't necessarily be visible. You could quantize as you're recording to get your notes more on-beat.
 
but use iterative quantise not hard quantising i.e. use a quantise that nudges notes towards the nearest grid division rather than pushing them onto that grid division

even a latency of 40ms is barely perceptible unless you are a high speed performance athlete (cycling/car racing/motorcycle racing)
 
When I play the drums with my hands then to myself it sounds like it would when they have gotten quantized, though when I have recorded the drums then they do for some reason sound off.

In my opinion the way it looks like is that the notes I am playing do get recorded a bit off and during playback they get played a bit off. Though, when I quantize the notes for example with drums, then it does not get noticeable as much, yet with more instruments it does not sound good anymore, also it is better to compose music without quantize.

EDIT:
Does somebody know if I would be seeing better results with an AMD Athlon 5350 (4 x 2.05 GHz)?
 
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If you mean "no pops and clicks" - You need a properly configured system with well-optimized drivers.

A very powerful PC system can still have DPC Latency issues caused by driver conflicts/incompatibility, poor hardware seating, insufficient power, power-saving options, etc.

If you download Resplendence LatencyMon on Windows, you can tell whether your computer is configured properly for low-latency audio:



Aside from that, you just need to make sure that you have a system that's suited to size and style of your projects.

For example, if you use a lot of Kontakt (which generally uses DFD "Direct From Disk" to stream sample tails), you'll benefit from an SSD more than someone who does not.
If you swap it out of DFD mode and into Sampler mode, you'll benefit greatly from having a lot of RAM.

It really just kinda depends.

I can help make some suggestions to make the best bang for your buck if I understand your usage cases,
but until then, it's hard to make suggestions.

-Ki
Salem Beats
 
With my system it said I should disable "CPU throttling".

Salem Beats, thanks for the post.
In regards to the screenshot you have posted, what type of system do you have? Can you let me know please.
 
ftxn said:
With my system it said I should disable "CPU throttling".

Salem Beats, thanks for the post.
In regards to the screenshot you have posted, what type of system do you have? Can you let me know please.
Click to expand...

I have my system specs listed on my "Salem Beats Uses..." page.

However, the screenshot I posted isn't of my system.
Instead, it's just a stock photo I grabbed off the web.

My system runs a lot lower steady DPC latency than the screenshot:
About 20-80 microseconds with everything hooked up,
with a short 400 microsecond spike from the nVidia drivers once every 5 minutes or so.

I don't have a USB audio interface (I use a native internal PCIe card, as you can tell on my list), and not having one generally helps with DPC latency.
This is because with most systems, all USB devices end up being processed by the same hub, and each of the devices gets in one another's way.
When you have a USB audio card, the requests it submits to your CPU to fill its audio buffer can be interrupted by the mouse's requests to update cursor position, a USB keyboard's requests to enter key combinations, etc.

Speaking of DPC issues - I've had an issue in the past where simply replacing an (apparently innocent and working) USB cable fixed a DPC issue I was having with my USB drivers.
Although it appeared to work just fine on the surface, presumably it was partially damaged, and therefore needed to re-transmit some data packets, causing a lag on the system as the CPU needed to wait for these packets to be received before the system would allow it to work on anything else.
Hence, high-quality USB cables with no kinks are a surprising "must-have" for high performance in a computer that uses USB devices.
I don't buy those $2-for-4 USB cable packs from Amazon/eBay like I used to.

-Ki
Salem Beats
 
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Thanks for the help so far.

I was looking to ask if it matters, when I mention that I would have the same results with a desktop PC (similar specifications) and the same audio interface (Yamaha Audiogram 3 and its own ASIO drivers)?

Could the audio interface really be the issue, in comparison with which audio interface may I see better results?
 
ftxn said:
Thanks for the help so far.

I was looking to ask if it matters, when I mention that I would have the same results with a desktop PC (similar specifications) and the same audio interface (Yamaha Audiogram 3 and its own ASIO drivers)?

Could the audio interface really be the issue, in comparison with which audio interface may I see better results?
Click to expand...

I'm not saying that it is, but it definitely could be.
You can begin to troubleshoot by uninstalling its drivers, disconnecting it from your computer, and restarting.
If your DPC latencies continue to spike over acceptable levels after you've done this, the Audiogram drivers are not the problem (or, alternatively, they are a problem, but not your only one).

-Ki
Salem Beats
 
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