Jim Roseberry

The higher the sample-rate, the smaller amount of time for a given ASIO buffer size (which reduces latency). Using high sample-rates is one way to potentially mitigate higher round-trip latency. The downside; CPU use will be considerably higher Note that the audio interface's safety-buffer is the major X-Factor when it comes to round-trip latency.

+1 on the dedicated audio interface. Integrated audio was never built for use with DAWs. The drivers are not robust... and don't provide low-latency. The audio inputs/outputs are also unbalanced (resulting in more noise).

The noise-floor on that pedal-board would have to sound like a jet-plane. 😂

FWIW, That DPC Latency is unfortunately on the very high side. Working at low-latency isn't going to happen without glitches. Even working at mid latency settings is going to be tough when DPC Latency is upwards of 1000uSec. Peak DPC Latency for a Clevo shell laptop should be ~200uSec.

The only advantage to 500 series gear is the small form-factor. You can put several pieces of "a-la-cart" quality gear in a small space. Kind of like a guitarist's "pedal-board" ... but for studio gear. Rupert Neve will tell you a 500 series unit (due to size limits - smaller PS and Transformers) is not equal to the full sized option.

If your machine has Thunderbolt-3, ~$600 would get you into Quantum-2. Quantum can get RTL down to 1ms. If you need to go USB and want something simple, have a look at the Audient ID14 (~$299). Best fidelity you'll find at that cost (Burr-Brown Converters) Round-trip latency is ~5ms at a 64-sample ASIO buffer size 44.1k To achieve better than ~5ms total round-trip latency with a USB audio interface, you're looking at RME or MOTU (recent AVB series and spin-offs). Either of those will get you down to just under 4ms RTL.

Back then, having good sounding EQ and Dynamics on every Channel/Bus... along with quality Delay/Reverb was a real break-through. It's funny how we (now) take all that for granted.

FWIW, To my knowledge... all the new audio interfaces that have USB-C ports are actually USB-2 (not USB-3.1). There's zero performance advantage. Some of these new audio interfaces come with a USB-C to USB-2 adapter cable (to connect to standard USB-2 ports).

Sample-rate refers to the number of samples per second 44.1k = 44,100 samples per second 48k = 48,000 samples per second 96k = 96,000 samples per second The higher the sample-rate, the quicker a given buffer size. 64-sample ASIO buffer size at 44.1k = 1.5ms 64-sample ASIO buffer size at 48k = 1.3ms 64-sample ASIO buffer size at 96k = 0.67ms Thus at a given buffer size (64-samples in this example), as sample-rate gets higher... latency gets lower. ie: Doubling the sample-rate (twice as many samples per minute) cuts the latency of a buffer in half. One might then expect that doubling the sample-rate would cut your audio interface's round-trip latency in half. The X-Factor is the driver's safety-buffer. If the driver's safety-buffer (often hidden) is consistent size across all sample-rates, doubling the sample-rate will cut round-trip latency in half. If the driver uses a larger safety-buffer for higher sample-rates, round-trip latency will be slightly lower at higher sample-rates. Many audio interfaces don't allow using ASIO buffer sizes smaller than 64-samples when working at sample-rates above 48k.

To my knowledge, all the latest audio interfaces that have USB-C port (Focusrite, Presonus, M-Audio, etc) are actually USB-2 (not USB-3.1 like you might expect). This offers no performance advantage vs. connecting via USB-2.

FWIW, You don't need closed-loop water-cooling for the 8700k. You can use a large quality air-cooler... and it'll run quieter. BTW, You can easily lock all six cores at 4.7GHz (no increased noise or stability issues).

I had two of the DSP Factory cards. Another great piece of hardware from days gone by...

I believe that's the outboard PCIe to PCI adapter that's been working with the Echo cards... but Search the forum to make sure it's the exact same model.

Read section 6.5 http://tldp.org/LDP/tlk/dd/pci.html

Bridged PCI slots... To my knowledge, no current generation motherboard has native PCI slots. The OP's best course of action is to get a quality Z370/Z390 motherboard... and use the outboard PCIe to PCI adapter that several here have working with Echo cards. It's a zero risk choice. FWIW, You really don't want to let 15 year old hardware determine motherboard choice.

If you're running a B75 chipset motherboard, it has native PCI (not bridged). The Echo cards don't cope well with bridged PCI slots.

FWIW, That's a bridged PCI slot. Echo PCI audio interfaces are notorious for *not* working in bridged PCI slots. In fact, we've never seen an Echo card work in any motherboard with bridged PCI slots. There are folks here successfully using an outboard PCIe to PCI adapter with the Echo cards. IMO, The OP would be much better off getting a quality Z370/Z390 motherboard (Coffee Lake CPU)... and getting the outboard PCIe to PCI adapter that's known to work.

Shipping to the UK is expensive. We built a custom machine for UK based composer Evan Jolly (Hacksaw Ridge, Wonder Woman, etc). Shipping to the UK was ~$600

Yep. That's a bit of a pain... even if you have the spare CPU.

FWIW, 550w power-supply is on the lean side for that build. It may work fine... depending on the number of drives, bus-powered USB devices, etc.

If you can get a good deal on it, I'd go with the i9-9900k (8 cores, 16 processing threads). All cores can be locked at 5GHz. With the right cooler, it runs near dead silent. Regarding Asus vs. Gigabyte, the answer is "yes". 😉 We've used many Z370/Z390 motherboards from both... all with reliable performance. Get the board that has the features you want.

When it comes to DAW related advice: I've been around since the days when the Cakewalk "forums" (News Groups back then) were on CompuServe. Few folks have accumulated more hours/experience building/supporting DAWs. There's nothing I post that can be construed as bad advice... or that's going to leave someone stuck/non-functional. Folks contact me on a daily basis to avoid/eliminate issues. With the right set-up, software-based monitoring is practical/usable in the here/now. It takes the right audio interface, it takes a fast machine, and it helps if the software has Hybrid Buffering For me, having ultra low latency makes all processes more enjoyable. Playing virtual-instruments - timing/response is tight Triggering drum samples - timing/response is tight Playing/monitoring in realtime thru AmpSim plugins - timing/response is tight With such low latency, you've got the feel of hardware-based monitoring... with all the flexibility of software-based monitoring.

Ironically, I'm a singer when performing here in Columbus, OH. 😉 Keep in mind that I've been building DAWs professionally for 25 years. My machine is running a 9900k (8 cores, 16 processing threads all locked at 5GHz). Even with Quantum set to a 128-sample ASIO buffer size 44.1k, total round-trip latency is 2.9ms. With a DAW that's using "Hybrid Buffering"; even on dense projects... I'm not going to run out of processing power for software-based monitoring. I used a combination of hardware and software based monitoring for most of the last 25 years. 😉 I've been on sessions in Nashville (Memphis Horns, Terry McMillan, Tabitha Fair) decades ago where we had to monitor straight off the console (to avoid latency issues). At those sessions, I was on the phone with Charlie from Frontier Design (soldering iron in hand) to mod their Dakota PCI card to achieve sample-accurate sync. Until recently, *effective* software-based monitoring wasn't practical. With today's hardware (Quantum and similar) and machines with super high clock-speed, software-based monitoring is practical/effective. ie: If you're someone who's using V-Drums to trigger sample libraries like Superior Drummer 3, BFD3, etc... it's nice to be able to do that at super low latency. Can you work without software-based monitoring? Of course! I've been working around it for decades. Having super low round-trip latency benefits numerous facets of the production process... and opens new possibilities.

While it's on my mind... Another example how monitoring thru software can open up new possibilities: I've got one of the new HeadRush Gigboard guitar processors. I've programmed a nice Marshall JCM-800 patch (with various optional boosts/EFX)... and I'd like to play/hear that in realtime with stereo Cab IRs (not just a single mono Cab). HeadRush can run a pair of Cab IRs... but that (along with a single Amp) nearly maxes out its DSP. With the ability to monitor thru software with 1ms total round-trip latency, I can set-up a stereo pair of Cab IRs in my DAW software. Also, I can set-up pristine Reverb and Delay plugins... and hear the guitar thru all the above in realtime (as I'm playing)... without latency issues. This is very flexible... as you can swap Cab IRs... and adjust Reverb/Delay at any point. The new AxeFX III lets you mix up to four simultaneous Cab IRs in each of two separate Cab Blocks. The above method could be used to yield similar results. With new capabilities come new opportunities...

Hi Clint, The AudioBox VSL reports latency accurately. 4.9ms total round-trip latency at a 64-sample ASIO buffer size 44.1k