Jim Roseberry

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

With the "Hybrid Buffering" approach used by Studio One (Logic and Samplitude had this years ago, ProTools got it at version 11), you don't have the big CPU hit from monitoring thru software. Tracks that are merely playing back are processed using a much larger buffer size. Tracks that are being monitored thru software are processed using the small buffer (only while being monitored). This gives you the best of both worlds. When you can monitor thru software (no glitches) at 1ms round-trip latency, it pretty much makes hardware based monitoring moot. You can monitor thru AmpSim plugins in realtime... or anything type of processing... and never give it a second thought. When you go to play a virtual-instrument... one-way Playback latency is incredibly low.

Some audio interfaces don't report their record latency accurately. Check the Record Offset for your audio interface. Record a short high-transient spike signal (like an isolated click). Now, take that single "Click" and rerecord it (physically patch an output to an input) to a second track. Zoom way in... and measure the difference between the two clicks (in samples). This is your audio interface's Record Offset. In Sonar, under Preferences>Audio>Sync And Caching... enter the number of samples (Record Offset) in the "Manual Offset" box. Newly recorded audio will now line up correctly.

The Roland audio interfaces have some nice features. Lowest possible round-trip latency just isn't their forte'. At a 48-sample ASIO buffer size 44.1k, round-trip latency is 7.4ms.

Rendering video makes even a large/dense audio project look light-weight. IME, The ASIO implementation in Premier, After-Effects, etc is not quite on-par with better audio applications.

A 64-sample ASIO buffer at 44.1k = 1.5ms (This is true no matter what audio interface you're using) While true (and I know you already know this), that's not telling the whole story. When you're monitoring in realtime thru software EFX/processing, you're dealing with round-trip latency: ASIO input buffer (1.5ms) ASIO output buffer (1.5ms) A/D D/A (~1ms) Driver's safety-buffer - this is the X-Factor when it comes to round-trip latency and it's often hidden (can vary radically) In this example, we're already at 4ms... without factoring in the safety-buffer. If the audio interface is one of the better makes, the safety-buffer will be small and round-trip latency will be ~5-6ms. If the safety-buffer is large, round-trip latency can be more than double.

In the case of playing soft-synths, you're dealing with one-way (playback) latency. Monitoring audio tracks in realtime thru software EFX/processing, you're dealing with full round-trip latency. The UR44 (IIRC) yields about 7ms total round-trip latency at a 32-sample ASIO buffer size 44.1k. Some of the absolute latest generation of AmpSim plugins are sounding pretty decent. Helix Native sound pretty good. The new version of TH-U (from Overloud) is going to have a feature similar to Kemper's ability to "profile" actual mic'd amps/cabs. PRS Super Models sounds good (IMO) if you use different Cab IRs Onboard DSP to process/route/mix/loop-back-record can be extremely useful (if you use it). If you're after lowest possible round-trip latency, onboard DSP will slightly increase it. Part of the reason Quantum can achieve such low round-trip latency; it has zero onboard DSP for routing/mixing/loop-back-recording. All monitoring has to be done via software. Why so fixated on lowest possible round-trip latency? In the case of Quantum, since all monitoring has to be done via software, it's critical. Lets say you're using a Kemper Profiling amp... or something like Helix or HeadRush (all hardware guitar amp sims). The Kemper itself can have up to 4ms round-trip latency. The whole point of an audio interface like Quantum is to keep round-trip latency as low as possible. If Quantum were yielding 4ms latency, add the Kemper's 4ms latency... and you're at 8ms total round-trip latency (while monitoring guitar). That's a significant step backward compared to hardware based monitoring (8ms vs. near zero). Since Quantum can actually get down to 1ms total round-trip latency, even with Kemper's worst case scenario, you're at 5ms total round-trip latency. At 1ms total round-trip latency, Quantum makes software based monitoring effectively on-par with hardware. Monitoring via software at 1ms round-trip latency hits the CPU hard. High CPU clock-speed is critical... as this isn't a process well-suited for multi-threading.

In today's economic world, few have the luxury of being 100% altruistic.

The 896mk3 Hybrid interface (and all the Hybrid series) was (round-trip latency wise) a step backward from the original 896HD. The original 896HD yielded 5ms round-trip latency at a 64-sample ASIO buffer size 44.1k The Hybrid series added onboard DSP processing/mixing... and that increased round-trip latency to ~6.5ms at those same settings. To get sub 4ms round-trip latency from MOTU USB, you have to be running one of the newer AVB models (or spin-offs). The newer MOTU USB drivers allow you to tweak the safety-buffer size. FWIW, Thunderbolt under Windows is not a crap-shoot. You just have to make sure you've covered all the details. MOTU was one of the first companies to have (release version - not beta) Thunderbolt drivers for Windows that support "PCIe via Thunderbolt" (allowing PCIe level performance).

The reason they gave you that advice... Echo cards never worked with bridged PCI slots on motherboards. Lynx cards also had major issues with bridged PCI slots. RME and M-Audio were much more forgiving... and worked with most bridged PCI slots.

Add an instance of Kontakt (and load the desired sound/s) Add an Audio Track Assign the Audio Track to receive audio from that Kontakt instance (Input drop-down list) Assign the MIDI track to output to the instance of Kontakt (Output drop-down list)

This doesn't answer the OP's question... but he might want to check out Melodyne (light version included with CbB). Melodyne is far more advanced (sounds better) for intonation adjustment.

Well placed overheads can capture the bulk of a good drum-kit sound. This is a great example of that... Well done

+1 scook Open the type 1 SMF... and immediately save it as a Cakewalk Project. Often times, there are unwanted Tempo, Patch Changes, and Volume/Pan MIDI events. I'd strip those out unless specifically desired. Then, assign the individual MIDI tracks to specific virtual or hardware instruments.

Delete CbB's "aud.ini" file. You'll have to reset your audio preferences. In similar cases, that often resolves the issue.