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Jim Roseberry

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Everything posted by Jim Roseberry

  1. FWIW, We've seen the Fast Startup option (when dual booting) cause loss of data files. If you're dual-booting (even two copies of Win10), make sure it's disabled.
  2. Do you have any MIDI interfaces or USB connected MIDI controllers? This can happen if a driver doesn't let go. A good while back, this would happen if you had a Korg keyboard or MIDI controller. I believe the workaround was to turn off the Keyboard/MIDI controller, then close Sonar (CbB).
  3. I hear what you're saying about Win10 and what I refer to as the "annoying components"... but those things can be reined-in with the Pro version. Once Cortana, OneDrive, all automatic updates including notifications, privacy settings, etc have been disabled... Win10 is a fine DAW platform. FWIW, I feel like it's more of a "dumbing down" of the OS (trying to make it easy for the less tech-savvy) than a matter of control. And in that, it's two-steps-forward... one-step-back. On a computer, I rarely like anything "automatic".
  4. Who said the Presonus Quantum was a Thunderbolt-3 audio interface??? I said it was connected via Thunderbolt-3. "PCIe via Thunderbolt" (necessary for PCIe level performance) is only supported via Thunderbolt-3 controllers (see below). To clarify: Microsoft absolutely DOES support "PCIe via Thunderbolt", but only on Windows 10 and only with Thunderbolt-3 controllers. USB-C is the type of Port that's used to carry Thunderbolt USB-C can also be used to carry USB-3.1 Gen-2 Just because a motherboard has a USB-C port/s does not mean it has Thunderbolt or supports it. The motherboard has to have a Thunderbolt-3 controller onboard... or it has to have a Thunderbolt-3 header and BIOS support for a Thunderbolt-3 AIC (add-in-controller card). I'm well aware of stage latency (when not using IEMs). Yes, you can compensate/anticipate... but walk about 50-100 feet off stage into the crowd. I play the better part of 50 shows a year locally. And no, I don't monitor our guitar players sound from his amp that's 20' away. It's mic'd and the sound sent to my IEMs or wedge (much lower latency). I gave specific examples of how anyone can test and see/hear for themselves that lower latency is indeed noticeable and feels more immediate/tight. Take that 12+ms round-trip latency into a session with any seasoned player... and I guarantee they're going to hear it... and not like it. Been there and done that in Nashville way back in the day (recording the Memphis Horns, Terry McMillan, Tabitha Fair, etc). Even the vocalists noticed the latency... and complained about it. The solution back then was to split the signal and monitor off the board... so the performers could hear exactly what they wanted/needed (tone/EFX wise)... and not have the latency impede their performance. If you can't feel the lag of latency until it gets to 30ms, more power to you. I guarantee that's not the case with many folks. I talk to them every day. A composer using virtual-instruments to score for TV/Film doesn't want to play thru 30ms latency. When one of our clients has Slash over to overdub guitars, he doesn't want to cope with 30ms of latency while monitoring and trying to play tight. Fidelity comparison of Quantum vs. an audio interface that's half the cost? Which do you think will have lower noise-floor, more stable digital-clock, quieter mic preamps, more dynamic range on the A/D D/A? BTW, You can buy an audio interface that has better fidelity than the Quantum... but you'll spend a lot more to get it. The Quantum is (right now) one of the best low latency performers. That's why I moved from an Apollo 8 Quad and Quad Expander (lowest round-trip latency is about 3.7ms). As a "power-user", if you can't feel the lag of 12+ms latency (especially on high-transient material), I don't know what to tell you. It's as easy to loading your favorite piano sample library... and playing at each ASIO buffer size. The difference is obvious and immediate. I was not talking about MIDI jitter/slop. That's another conversation. As you know, MIDI is a serial protocol (not parallel) and doesn't lend itself well to ultra-tight timing. Newer MPE controllers that spit out a lot of CC data can cause MIDI data stream congestion (causing timing issues). One reason that folks like virtual-instruments; once the MIDI part is captured, playback timing is sample-accurate (consistent)... unlike dealing with hardware instruments. I had a Korg O1/W FD years ago that had an onboard sequencer that was all but useless. If you tried to record joystick movements, it would clog the MIDI stream causing major timing issues. Not much of a workstation if you can't effectively record. There's a time/place for working at higher sample-rates. It's not practical all the time. It's overly simplifying the situation to say that humans can't hear the difference when recording at 44.1k vs. 88.2k and higher. It's not just about hearing higher frequencies. For things like AmpSim plugins (dealing with non-linearity), you can have aliasing happening in the audible range. Perform the distortion processing at high sample-rates, and the aliasing is above human hearing range. Even when converted back down to 44.1k, there can be a clear audible improvement. Some plugins up-sample... as well as some DAW applications (CbB has an up-sampling option). BTW, I wasn't a huge Presonus audio interface fan... until they released Quantum. Had always used RME or MOTU, then moved the UA Apollo to take advantage of their "Unison" technology (monitoring UAD plugins with 2ms round-trip latency), then moved to Quantum due to its incredible ultra low-latency performance. Presonus has a rock-solid performer in Quantum. I should mention that Quantum can actually go down to a 16-sample ASIO buffer size. I didn't mention it previously because no current machine can effectively make use of it. Still, I'd rather have the option and not (currently) be able to use it. Eventually, machines will be fast enough... and 16-sample ASIO buffer size will be usable. Note: 64-sample ASIO buffer size at 44.1k = 1.5ms 32-sample ASIO buffer size at 44.1k = 0.75ms 16-sample ASIO buffer size at 44.1k = 0.375ms Round-trip latency is the sum of the following: ASIO input buffer ASIO output buffer The driver's (often hidden) safety-bufferer A/D D/A When it comes to round-trip latency, the audio interface's safety-buffer is the X-factor. The best audio interfaces can use a small safety-buffer... whereas other audio interface's have to use a large safety-buffer. The smaller the buffer, the lower the latency. If I'm overdubbing drums (and trying for tightest timing), I'm certainly not going to listen to a monitor that's placed 12 feet away (~12ms latency). I'm going to use IEMs or headphones. There's absolutely nothing wrong with your Focusrite audio interface. There's also nothing wrong with acknowledging that there are other audio interfaces out there that are better performers for those who are concerned with ultra low latency.
  5. You can migrate the OS install to the new machine... but it'll be a mess. You're much better off starting with a clean install... and taking the time (when finished) to create an up-to-date backup image file. This way, you're starting on a rock-solid clean/lean foundation... and you've got a backup of the clean install.
  6. "Except by default, a 32-bit user mode process can only access 2GB." It's a whopping 3GB with the Large Address Aware switch enabled. 😉 That takes us back a few years...
  7. Exactly. I use Drop-Box on an "as needed" basis. But I don't want it (or any other application) automatically trying to backup my files (especially not audio/video files).
  8. A 32Bit application can only address 4GB RAM. If you're working with virtual-instruments, that's a tough limitation. ie: Loading a full kit in Superior Drummer 3 takes about 2GB RAM. Development resources are finite. In today's economic environment, it doesn't make sense for a company to spend resources on antiquated products. BTW, CbB is not alone in this situation. Steinberg doesn't make a 32Bit version of Cubase 10 Line-6 only makes a 64Bit version of Helix Native
  9. I use the Adobe Creative Cloud (good use from it), but they are the worst at having applets running in the background. Even with services set to "manual start", you have to manually stop creative cloud processes. They won't stop upon closing Photoshop, Premier, After Effects, Audition, etc. I have the same experience with Bandlab Assistant. 😉 Granted, it's not used often
  10. Both Microsoft and Apple have caved to the "average" user. Apple has all but abandoned their power-users. Look at their machines... virtually no user-serviceable (internal) expansion A new boot drive means a trip to the Apple store There's no expanding RAM for any current MacBook Pro (what you buy is what you get - WYBIWYG) Lucky for us, with the Pro version of Windows 10, we can still take full control.
  11. I'm not sure if the above was serious... or in jest By today's standards, running 6ms total round-trip latency isn't really that low. ie: RME Fireface-800 (now about 15 years old) connects via Firewire... and can achieve 6ms total round-trip latency at 44.1k using a 64-sample ASIO buffer size. Using a Presonus Quantum (connected via Thunderbolt-3), you can do things like run Helix Native (software only version of the Helix guitar processor) at 96k using a 32-sample ASIO buffer size. That's 1ms total round-trip latency. To be able to to this effectively requires a fast well-configured machine. If the comments about latency perception were serious, you absolutely can tell the difference between 6ms and 12ms latency (especially with high transient instruments). Dial up your favorite piano sample library... and set your ASIO buffer size such that playback (one way) latency is 6ms. Now double the current ASIO buffer size so that playback (one way) latency is 12ms. Which one feels tighter and more immediate while playing? If your audio interface permits, cut the original ASIO buffer size in half (3ms playback latency)... and compare playing response to the previous two settings. It's pretty easy to feel the difference. If you're going to monitor thru software: Anything higher than about 6ms total round-trip latency starts to feel pretty sluggish Upwards of 10ms total round-trip latency starts to feel unbearably sluggish Below 5ms starts to feel more comfortable Below 3ms feels tight If you have an audio interface that allows super low round-trip latency, you can test this for yourself. Set the audio interface to operate a 1ms total round-trip latency Open a test project with a single audio track Insert a delay plugin that offers fine control (1ms increments) and set the delay to be 100% wet Use the delay plugin's time parameter to simulate various amounts of (additional) round-trip latency If you're going to do something like use a V-Drum kit to trigger drum samples from Superior Drummer 3 (in realtime), you're going to want the lowest possible round-trip latency. Round-trip latency above about 3ms will start to affect the feel. Round-trip latency of 6+ms and the drummer will feel he's playing thru molasses. With round-trip latency below 3ms, the drummer will feel much more comfortable.
  12. Assuming the OP is running a desktop/tower, there's numerous solutions. ie: You can get a single 5.25" bay that has four removable 2.5" trays for SSDs (for SSDs in a single 5.25" bay). If the OP is running a laptop or small form-factor machine, there may be zero internal expansion. If the OP is running a laptop/SFF machine that has Thunderbolt-3, using an external Thunderbolt drive enclosure is also an option (albeit somewhat expensive).
  13. Most docking stations (connected via USB) have firmware that puts drives to sleep after a brief period of inactivity. Unfortunately, there's no means of disabling this "sleep" feature. External eSATA enclosures typically don't put the drives to sleep. If you're going to use a conventional HD for projects, I'd go with a 3.5" drive (higher performance). If you' going to use SSD, it's a moot point.
  14. I've been using Acronis True Image for going on two decades. Have always used it the same way (manually booting from one of their "Rescue Discs") I've never experienced a failure to backup/restore properly. The beauty is the operation is completely clean (nothing installed/nothing running in the background that needs to be shut down)... and the process is done completely outside of the OS. Macrium Reflect and Paragon Backup & Recovery are both good choices. If you install backup software, make sure to disable scheduling services/etc.
  15. Yes, project data on the boot drive is a bad idea for several reasons. For backup purposes, you want to keep the boot drive clean/lean. Otherwise, backup is slow/tedious... and consumes a lot of additional (redundant) space. For top performance, you want the audio files streaming from a separate drive. If the boot drive were to die prematurely, your project data is still intact on the data drive/s.
  16. It's completely safe to delete all contents of the picture cache. The files will be recreated as needed...
  17. Also wanted to chime in on the multiple DAW applications vs. CbB With what I do for a living, I have all the major DAW applications. Each has incredible strengths... and head-scratching weaknesses For straight up audio work, both Reaper and Samplitude are incredibly powerful (both having Item/Object based editing/processing). For more advanced MIDI, it's hard to beat Cubase. Almost all of our professional composer clients are running Cubase. Studio One is extremely easy to use... and has a nice balance of features/performance. However, it lacks in more esoteric MIDI features. ie: Percentage Quantize is limited to 50%. Why not let the user choose the desired percentage??? ProTools 2018 offers a well balanced feature set, but CPU efficiency (especially when working at the smallest ASIO buffer size) isn't as good as Reaper/StudioOne/CbB. When it comes to CPU efficiency, Reaper is the top performer. What's Reaper's weakness? It's configurable almost to a fault. Initial configuration can be daunting... especially for less tech-savvy users. More esoteric MIDI features are lacking compared to Cubase. Ableton Live is fantastic for working with samples, triggering virtual-instruments/samples (especially live on stage), etc. It's weakness is on the editing side (lacks many advanced audio editing features found in Reaper/Samplitude, SO4, CbB, Cubase, ProTools). Even with numerous (good) options available, I'm not 100% settled on which application will be my main DAW software. Of late, most of my time has been spent with Studio One. There's a lot to like about SO4, but there's also a lot to like about CbB. Truth be told, most of us could make do with any of the above. That's when it starts to feel like we're spoiled by so many quality choices.
  18. Wanted to chime in on "Dropped Buffers" Dropped buffers aren't because of your audio interface or the A/D D/A. If buffers are lost/dropped, it's because the machine can't keep up with the sustained data-flow. High DPC Latency is often a culprit
  19. Companies like Fractal Audio and Kemper raised the bar for guitar (amp-sim) processors. This caused other companies like Line-6, Atomic Amps, HeadRush (InMusic), UA, etc to step-up their modeling. Line-6 (IMO) nailed a great UI with Helix. Super easy to use. They also released Helix Native (software only version of Helix), which is extremely convenient. I've owned/used all the major guitar modeling/profiling processors... and compared most side-by-side. In short, ALL are capable of good/excellent results. All are also capable of sounding bad. The more familiar you are with your chosen modeler/profiler (and the original gear), the better your end result.
  20. A clean install means reinstalling all software/plugins (reauthorizing everything). This is one case where using dongles is actually more convenient. Your license is on the dongle, so there's no re-authorization when reinstalling.
  21. By today's standards, 16GB RAM for a DAW isn't a "stretch"... it's common place. If you're running 8GB RAM (especially using virtual-instruments), that's running pretty lean (even with Win7). The OS itself is going to take some of that 8GB (Win10 with Chrome open to write this message is using about 2.8GB). So, even in your Win7 scenario, say the OS is taking 2GB of the 8GB total RAM. That leaves ~6GB before the machine starts to hit the VM swapfile in lieu of real physical RAM (which will absolutely kill performance). If you're running something like Superior Drummer (where the samples are loaded/streamed from RAM), that can consume another 2+GB. You're now down to ~4GB for your DAW software and any remaining plugins. Even if it's working fine, that's definitely running lean. It's not about doubling/quadrupling your RAM for Win10, it's about having enough RAM for your largest projects (to avoid hitting the VM swapfile). The doubling/quadrupling is the reality of having to install dual-channel (two matching sticks) for maximum performance. With each OS release, the OS is slightly larger... and the system requirements creep up. Been the case with Win95, Win98, WinNT, Win2k, Vista, Win7, and Win10 One could say each OS becomes slightly more "bloated" compared to the previous version. Hardware speed/capacity increases... and the OS/Applications/Plugins are developed to use this increase in speed/capacity. If you were transitioning from Win98 to Vista or Win7, you'd be feeling exactly the same. Win10 is a fine DAW platform. Increase your RAM to 16GB and tweak the OS for maximum DAW performance. If you want the option to run Thunderbolt, you have to be running Win10. If you want to run the latest generation of hardware (ie: Z390 chipset motherboards), you have to run Win10. If you want to run the latest DAW software/plugins (smoothly), it makes sense to run the latest OS (not one that's a decade old). Some software developers have stopped official support for Win7 (more will follow). That's a reality of finite development resources.
  22. In the context of a dense Rock mix, a typical DI electric bass (especially a passive bass) is going to sound a bit anemic. If you have access to something like a Neve preamp, that can help immensely. The sound is larger/smoother (without sounding compressed). I struggled for many years to get a good DI electric bass recording (especially with passive basses). I used the Avalon U5, Reddi Box, UA-610... and all were OK sounding (to my ears)... but not great. Ultimately got a Neve Portico-II channel-strip... and it was what I'd been looking for all those years. Though it has a great 4-band EQ and nice dynamics processor, the sound of the bass straight off the preamp sounds great. At mix, a very slight amount of compression and a very small bit of high-pass filter to roll-out the very deepest sub-bass The sound is there from the very beginning... If going DI with electric bass, keep in mind that a mic'd bass amp isn't going to reproduce sound all the way down to 20Hz. Use a high-pass filter to roll-out the deepest sub-bass (20-50Hz)... and the bass will sit better in the mix. If forced to use a "plain-jane" (for lack of a better word) type DI to record electric-bass, I'd use an Amp-Sim plugin to "toughen up" the signal. If you have a nice bass-amp and decent mic, consider mic'ing the amp. Sometimes it's quicker/easier to just record the real thing. If you have a great bass amp (Ampeg, Mesa, MarkBass, etc), I'd definitely try recording it.
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