Improving Push 2 Latency w/o an audio-interface

[Stromkraft]

The aforementioned Frontier Designs hardware which abandoned development didn't do so because of 'processors'. They simply went bust, as companies are sometimes known to do.

Yes, I use their Alphatrack in macOS 10.12 now. With a decade old drivers. Some people can write drivers that just will work, some cannot. Think Native Instruments for the latter.

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The cheapest interface I would recommend with good ASIO drivers in windows is the Steinberg Ur22.

I've been looking into this as an option, and noticed quite a bit of discussion surrounding problems people are having recently w/ audio dropouts and UR series interfaces

https://www.steinberg.net/forums/viewto ... 00#p692820

As recently as last month, many people say its still unresolved. They've released a number of driver updates (most recently Nov 30) and have another in beta-test. I'm curious: have you been using one of these interfaces yourself in a windows environment?

Are there any other models you think might have better-established drivers? I've heard mixed views about Focusrite gen1, but it seems like gen2 interfaces are better-received.

I had used various models with different computers, all worked fine but I have had a bit of problems in an old win7 system with a weak processor, but it should work fine IMO.

I have tried quite a few interfaces with windows and my first choice would be RME, stellar performance and stability but they are expensive units, the second best for me has been steinberg, at least they have support that are trying to solve the issues, focusrite and NI are very slow to update their windows drivers.

Supposedly the new Presonus quantumm line is good but they use thunderbolt and you will need a PCIe card for that I suppose.

[ceounicom]

focusrite...are very slow to update their windows drivers.

The selling-point for me was coming across this guy who points out that Focusrite's new (as of June 2017) Gen2 drivers work on all their legacy gen 1 hardware

http://masters-of-music.com/1st-gen-foc ... r-latency/

that means a big improvement in latency for legacy hardware that costs about 50% what the newer stuff does. For someone like me, that for the most part needs nothing but a headphone jack and a mix-cue, and doesn't care about 192K sample rates, is a good deal.

While i heard both the gen 1 and gen 2 scarlett series had some teething problems, they seem to have sussed most everything out by now.

by contrast - the problem identified in the Steinberg UR series (which seems concentrated around people with the X99 chipset, but is not exclusive to them, and may be a conflict with Nvidia display drivers) was first identified in spring of 2016, and they've still not figured it out entirely. They say they are dependent on Yamaha (who provided their driver) developers to fix the problem, which isn't encouraging.

[ceounicom]

my counter-argument was "why not just halve the buffer?". If the output latency goes down by an equal amount in 44.1 kHz when halving the buffer, compared to go 88.1 or 96, then I'd assume the machine would have to work less hard in the former case, so that would be preferred.

As i mentioned above, somewhere , i often tend to start projects with the lower buffer settings (e.g. 128-256), but depending on the VST instrument being used... more than 2 instances will require that to have to go higher in order to prevent glitching/stutter of playback.

the example i gave of "flat 512 buffer" was just a theoretical improvement in the latency for performance that i get *after* i've already passed that limit.

Meaning, i don't mind running hi-buffer settings at all...as long as all i'm doing is mixing/editing a project. The cost is nothing except an imperceptible delay with i hit "play".

But that same delay is completely unacceptable when i need to suddenly add a track and play a few new parts in, in real-time. So, solution: double the sample rate, which provides improvement in latency (while pushing the limits of the CPU)

i think it clear now the idea implicit in my headline-question for this post ("Improving latency w/o an audio interface") has a very simple answer: "You can't; at least not by all that much". I figured any answer would always include 'get an interface, dumbass', but i was probably naively hoping there were some tricks i hadn't yet considered.

thanks for all the feedback

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focusrite...are very slow to update their windows drivers.

The selling-point for me was coming across this guy who points out that Focusrite's new (as of June 2017) Gen2 drivers work on all their legacy gen 1 hardware

http://masters-of-music.com/1st-gen-foc ... r-latency/

that means a big improvement in latency for legacy hardware that costs about 50% what the newer stuff does. For someone like me, that for the most part needs nothing but a headphone jack and a mix-cue, and doesn't care about 192K sample rates, is a good deal.

While i heard both the gen 1 and gen 2 scarlett series had some teething problems, they seem to have sussed most everything out by now.

by contrast - the problem identified in the Steinberg UR series (which seems concentrated around people with the X99 chipset, but is not exclusive to them, and may be a conflict with Nvidia display drivers) was first identified in spring of 2016, and they've still not figured it out entirely. They say they are dependent on Yamaha (who provided their driver) developers to fix the problem, which isn't encouraging.

In the end I really just have my experience to share with you, I think the correct way to proceed for you is to try various audio interfaces, that's the only way you will know how each one performs with your system. Buying/selling secon hand can help you going through that, or finding an online store that allows returns.

[Stromkraft]

But that same delay is completely unacceptable when i need to suddenly add a track and play a few new parts in, in real-time. So, solution: double the sample rate, which provides improvement in latency (while pushing the limits of the CPU)

Yes, that was the insight I had on your problem. I was so used seeing the latency almost double with doubling the buffer that I erroneously assumed this was always so. I can't say I've noticed it much in practical work when using native audio, but usually when I don't have my Babyface, I'm not doing much recording, so the worse latency isn't as much of a problem when mixing something.

[ceounicom]

In case anyone happens to be browsing this now-dead thread:

what i ended up doing was getting a Focusrite Scarlett 2i4 (gen1) used on Ebay for about $70;

Focusrite updated their Windows drivers in June, and now both Gen1 and Gen2 hardware use the same USB driver-software. reports have been that they are vastly more stable (from 2013-2016 there were constant complaints, but it seems the last 2 iterations have worked out the major bugs)

The only added appeal this particular model had for me over other options was the fact that it is "DJ friendly": it has 2 pairs of aux line-outs (rca) aside from the main 1/4" stereo outs, as well as front panel switching for headphone monitoring between either 1/2 or 3/4 - basically, the bare-minimum you would need if you wanted to use it by itself for mobile DJ'ing. I also have hardware DJ mixers, but i liked the fact that, if necessary, i could probably use it as a cue-mixer. Other cheapo USB interfaces offer similar i/o options, but very few seemed to combine them with the ability to quickly monitor different stereo outputs independently.

I've been testing it for the last 2 days and the improvement in latency (and audio quality) is dramatic. I am running it with the buffer @ 64 samples, 48khz, and getting about 10-12ms total (in+out) latency. I have run test sets of 10-12 tracks with resource heavy VSTs and lots of fx/verbs/delays and i don't get any audio artifacts or glitches or dropouts; the CPU even seems to be taxed less - i am getting about 30-50% CPU usage in cases where i'd sometimes see it peaking higher (70% or so).

Overall i'm pleased, so far (god willing, i will not see the sort of crash-happiness of other windows-focusrite users). While people have rightly pointed out that this wasn't really a "Push" issue to begin with, the fact is that i'm now able to utilize the Push's super-sensitive pads in the way they were intended. Its nice to be able to whack at it with big, multi-voiced vst's and get instant feedback and zero audible latency or dropouts.

Thanks to everyone who chimed in with ideas and opinions. hope other people might benefit from my experience.