Hi everyone,
I've been running into an issue lately with Ableton Live 9 and my audio interface Traktor Audio 2:

Once I have a lot of Kontakt instances in my project (+EQ, reverb, etc.), I start getting crackles in the sound. It wasn't the RAM nor the CPU. After looking for the cause, I found that changing the buffer size to 1024 samples fixed the issue. No more cracks and pops...

However, I now get 30ms latency on midi input, which is a little hard to play to say the least (I had 18ms with 512 samples, which might seem high but it's actually quite playable).

So I'd like to know if any of you could recommend me a good audio interface that could handle a 1024 buffer size with an acceptable latency that's not too expensive (I just upgraded to 32gb of RAM for Kontakt I'm kind of broke right now )

I don't use external instruments, only vst's so I'm not interested in an audio interface with a gazillion inputs (could be zero), I'm really looking for a decent latency at a 1024 samples buffer size.

Thanks for the help *yes* ,

Cheers !

 

Hi zero-frag,

Can you give more infos about your config, in particular CPU type, motherboard brand/model and OS?

Sometimes, the crackles are due to high DPC latency, due to hardware or mobo tools (such AI Suite from ASUS). I'm using Focusrite Saffire Pro 14 (IEEE 1394 interface) and working fine at 32 samples 48 kHz (my projects are small, however, using REAPER with MIDI controller for live). Assuming 1024 samples is a very high value (maybe useful for big projects with many plugins).

Goggle "DPC latency", they're existing tools to test it (like DPC Latency Checker).

Sorry for English faults (I'm French).

 

Try to translate the information at this site, benchmarks for a bunch of different interfaces.

http://forum.dawbench.com/showthread.php?1548-Audio-Interface-Low-Latency-Performance-Data-Base

 

Sample buffer size is related to latency.

at 44.1khz there are 44100 samples in a second.

The theoretical lowest latency you can achieve with a 1024 buffer at 44.1khz is:

(1024 samples / 44100 samples/second) * 1000 msec/sec = 23.22 ms

Youre achieving 30ms. The 6.8 ms difference can be accounted for due to imperfect conditions and your plugins latencies. At best, with a new interface, you may be able to shave off 1 or 2 ms of latency which I highly doubt. So you're better off freezing tracks.

Despite what you have said, the culprit is the CPU its skipping calculations to keep up aka an under-run.

 

I knew that a higher buffer size would lead to a higher latency mathematically but I didn't know by how much. Thanks for clarifying that ! I guess you're right then, spending extra money on a high end audio interface would not be worth it. I might save 3 or 4 ms for a few hundred euros.

I thought maybe the CPU had nothing to do with it since load was at 80% but I guess it is related since reducing the number of samples gives more workload to the cpu, by definition. I might overclock my CPU a bit, I have a bunch of good coolers, so I can do it safely to an extent.

I guess I'll just have to freeze tracks like you say. Thanks for the reply, you saved me a few hundred euros

 

Thanks for the info, I actually have AI Suite from ASUS so that might be the issue, I'll check it out ! Merci !

PS edit: Thanks for the tip ! I tried the tool and found that I had a USB 3.0 driver that was causing DCP latency. Once disabled, I can now playback my heaviest project at 512 samples with no issue