Reducing latency in ableton...?

[Stevenglass]

 

 

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Here's my settings. I still have the smallest amount of latency and it's messing up my loop timing. I could edit and line it up but i'd rather get it perfect the first time. Anyone know any settings i could change? 12.7 is the lowest i get without fart noises.

Thanks

[WRGKMC]

latency can only be adjusted so low before you either get digital noise or dropouts.

Its dependant on the speed of your computer and how well the computers optimized.

10us is about as low as you can go with a high end quad processor with a good deal of memory.

the rest may be dependant on your interface. If you're using USB or firewire a realistic latency setting may be up to 100us.

You cant run effects in real time tracking. With playback and direct monotoring it doesnt matter how high your latency is, the program

buffers the playback so its in sync. PCI cards are the fastest because they tie into the main bus but even those have

processing delay and hard drives take time to read and write to the drive.

 

Run this tool. The average latency time is the lowest you can expect to get.

http://www.thesycon.de/deu/latency_check.shtml

 

If you get red spikes it means you have conflicts with other cards. Network and video cards

are usually the items that cause red spikes.

[Phil O'Keefe]

I'm not an Ableton user, so I can only help a little... if that.

 

What is the sound card / audio interface that you are using? Have you installed the latest factory drivers for that sound card / audio interface?

 

The input and output buffers being vastly different immediately caught my attention. Usually, buffer sizes are in multiples of 128, with a 128 buffer being what I usually try to use when tracking (for the lowest latency) and then after I finish tracking, I move up to a much larger buffer (1024 or 2048) to allow for less stress on the processor - you can usually run more processing and plugins at once with larger buffer settings.

 

I would recommend writing down the current settings (well, they're already in the screen shot... ) and then experimenting. Try 256 buffers for both input and output, and experiment with various combinations of the following values:

 

64

128

256

512

1024

2048

 

It sounds like 64 would probably be too low for your system for whatever reason, but 128 / 256 should work just as long as you're not putting huge demands on a slow processor with other stuff that's running simultaneously. If you're running a bunch of plugins or virtual instruments, that can have a serious detrimental effect on how low you can get the buffers (and latency) without it glitching.

[Phil O'Keefe]

Having had a second look, I noticed you're using a M-Audio Fast Track Pro, AND (more significantly) that the Driver Type you're currently using is MME / DirectX.

 

Change that to ASIO and you'll see a BIG improvement in performance!

 

After you try that, we can discuss the relative advantages and disadvantages to recording at 96kHz... personally I don't think it's really necessary for most projects, and it does place considerable additional demands on the computer's processor and requires more HDD storage space and throughput speed than recording at 24 bit / 44.1 kHz does.

[Stevenglass]

 

Having had a second look, I noticed you're using a M-Audio Fast Track Pro, AND (more significantly) that the Driver Type you're currently using is MME / DirectX. Change that to ASIO and you'll see a BIG improvement in performance! After you try that, we can discuss the relative advantages and disadvantages to recording at 96kHz... personally I don't think it's really necessary for most projects, and it does place considerable additional demands on the computer's processor and requires more HDD storage space and throughput speed than recording at 24 bit / 44.1 kHz does.

 

 

PHIL PHIL PHIL I LOVE YOU SO MUCH. OH MY GOD. IT JUST FIXED SO MANY PROBLEMS, THE SOUND WAS ONLY COMING OUT IN MONO, ETC. AAHHHHH I LOVE YOU.

 

 

So you'd recommend 24bit at 44k?

[Phil O'Keefe]

Cool - I figured that would help. MME is older technology, and ASIO is always the preferred driver format to use when given a choice.

 

And yes, in general, I feel 24 bit / 44.1kHz is fine. Not that there are not benefits to 96kHz, but for most things, I doubt most people will hear a difference, especially once everything's finished and then converted to an MP3. Like I said, there are also less demands on the system with 44.1kHz vs 96kHz, and unless you're doing a classical recording or acoustic instrument ensemble or something along those lines, I don't think 96kHz is worth the extra system overhead when the end improvement is probably going to be lost on the listener.

[Phil O'Keefe]

PS Do make sure you experiment with the buffer sizes - even under ASIO, you will want to increase them when mixing (for maximum CPU horsepower for effects, etc.) and use the smallest one you can run and still remain stable and glitch-free when tracking for the lowest possible latency. If your computer is reasonably modern and has a decent HDD (and you're running a session without a bunch of extra virtual instruments and plugins enabled), I would expect it to run 128 buffers without a hitch.

[Stevenglass]

 

PS Do make sure you experiment with the buffer sizes - even under ASIO, you will want to increase them when mixing (for maximum CPU horsepower for effects, etc.) and use the smallest one you can run and still remain stable and glitch-free when tracking for the lowest possible latency. If your computer is reasonably modern and has a decent HDD (and you're running a session without a bunch of extra virtual instruments and plugins enabled), I would expect it to run 128 buffers without a hitch.

 

 

Phil thanks for all the info and knowledge(i feel like i should be paying you haha). Thanks so much. I'll keep this thread bookmarked so i can refer back at any time. At the new asio settings the latency is 5ms. Which is way better the 12.7

[Phil O'Keefe]

At the new asio settings the latency is 5ms. Which is way better the 12.7

 

That's more like it, huh?

 

Let us know if / when you have more questions. We're here to help.

[Stevenglass]

At the new asio settings the latency is 5ms. Which is way better the 12.7 That's more like it, huh? Let us know if / when you have more questions. We're here to help.

 

Yes, the loops are timed perfectly now. So basically i'm just learning how to EQ my tracks. I think i've got the mic in the brightest position so now i'm going to EQ it the best i can.