mettelus

Flashing lights on hardware are typically codes for a fault. From this link on Focusrite's website this may be it: "However, if you find the gain or USB LEDs flash on and off constantly when your interface is connected, it usually means it is not able to draw enough power from your computer. Windows may also continue to attempt to connect the device, often denoted by a chime sound." When operating that interface, be sure you always have the laptop plugged in. Many laptops have inherent power saving mechanisms to conserve the battery if unplugged (like shutting off USB ports). Is the laptop plugged in and your power settings set to always on? The advanced power options in Windows are where you can surgically disable Windows from shutting off things, but some laptops also have their own proprietary software for power management... if yours has that, often there is a "disable when plugged in" option of some sort.

Also... track operations will be greyed out if nothing is selected. Unless I am missing something with that screenshot, a clip must be selected so CbB knows what it is splitting. Unless that theme shows selected clips as "grey," that clip is not selected.

Just to be clear, the "Generic ASIO" driver included with some DAWs is typically a variant of ASIO4ALL, and is actually a WDM driver that has an "ASIO wrapper" on it to fool applications into thinking they are using an ASIO driver (some applictions require an ASIO driver for output). The RealTek ASIO driver is faulty, and you can simply disable that in the Windows Sound Control Panel and be done with it. If you do not see this on the laptop, the program(s) which installed it on the PC would need to be reinstalled on the laptop. I am assuming Cubase installed the one you are using on the PC, but I do not use Cubase so cannot confirm. If Yamaha also has a driver, that would need to be installed as well from their driver software. CbB itself does not include a generic ASIO driver with it.

I think the OP is specifically referring to clip gain (where it will automatically redraw the wave form). Whether multiple clips are selected or not is immaterial (in fact, you can adjust clip gain even without the clip selected at all). CTLR-drag only works on the clip that is being hovered over.

Did you try shifting the driver mode to WASAPI or WDM and back to ASIO for that project? Reason I ask this is that it will force a reset on the sound engine drivers. CbB will save profiles of known good hardware and may be confused, so forcing that reset by shifting out of/back into ASIO mode will force CbB it to re-recognize the hardware available to it.

No problem! It is not quite the same describing something as letting you hear it. AmpleSound just does what the gpx file tells it (the two programs couple together well). Some engine features cannot be overridden (like its interpretation of picking, then hammer ons/pull offs on the same string) unless specifically told otherwise, which in turn is overridden by any performance details of the gpx file itself (some parts in the gpx file should have had "let ring" enabled, but they weren't and AS just blindly played it). This was more so you could hear the raw samples (the performance parameters/sound engine is another ball of wax).

Are is the reverb on a bus or in the track FX? Someone had asked about that video above a few months ago (using 2 busses, delay and reverb), and I had replied to that here. Using just one reverb bus, you will need: Post-fader out to Reverb Bus. Side chain input to the Soothe 2 on the Reverb Bus (used after Hyperspace in the bus FX bin, so the bus stays saturated).

@Brian Walton Roger that. Neither of those are good choices for what you are asking. I just did the following for this file. You can download and play with that and see how it measures up if you want to play with re-amping it. AmpleSound Peregrine Falcon (i.e., PRS), neck pickup. All FX are off, is set to DI mode (so the output can be run into another amp sim). The samples unto themselves are sort of boring, so I loaded in the "Hotel California.gpx" (Guitar Player) file that is on AmpleSound's site. Since they are played through the VSTi, it outputs in stereo. There are empty bars (6 bars after the intro) in the actual performance, so I deleted those. Basically, this is a DI performance of the gpx file, only the raw samples are used.

This will help for sure, but I think the OP is looking for something that adjusts real-time to the actual content of the track (at that instant). I use MSpectralDynamics as well, but it isn't cheaper (unless it hits a 50% off sale, then it would be).

Maybe I should ask for clarification on "legit clean," since to me "legit" is a sampled instrument, and "clean" is dry (or DI), and most would (or should) fit that bill. Am I confusing something here? AmpleSound is the one I am most familiar with, but you have to shut off each FX channel individually (all 4 of them) as well as the amp to hear the dry samples for each instrument. On the main screen itself there is also a "DI" toggle for "Sound Mode," as well as toggles for the doubler and fret sounds. Pretty much all of those are on by default (even for clean presets), so need to be shut off. Some VSTis like Heavier7Strings have a much more complex FX processing (some all over the face of the main screen), but even with that all of the samples in the sample library are dry for the pickup selection(s) used. There are just a whole lot more FX (and tweakable settings) in play with that one.

Unfortunately Sonitus Gate doesn't have any ratio control. I was sort of hoping that it was also an expander, but the Depth control on it seems to be more of a shelf with a step function gain reduction (why it can get flaky in use, especially with Depth set to -INF). I hadn't opened any Sonitus plugins in years, so the "free route only" was a nice challenge to that situation for me. There are some powerful freebies out there. That said, the Sonitus Multiband does have the ratio per band (and a highly tailor-able knee), and is actually a rather good choice for addressing low end and teaching compressors to new folks at the same time (since it has the display on the right so folks can watch the audio hitting that knee). The "Low" band can often address a good portion of what the Gate is doing (depending on the noise profile). Threads like this remind me of people wanting me to troubleshoot their car over the phone. Some things I just need to see/hear to address properly.

The step 2 basically is, and fairly open as to the "performance" part. MDrummer has one of the more massive sample libraries out there, so there may be similar for guitar coming down the pike. If they are working on a profiler, that would be icing on the cake (for me... I do have use for that). It is easy to speculate, but have to see what it materializes as. There are actually a lot of nice clean guitar libraries out there already (have to shut off all VSTi processing to check), but they tend to be 1) instrument specific (and often pricey) and 2) locked in a propriety format so you have to use the VSTi to play them (which is actually a good thing so it isn't just a "mess of files").

If you are able to post a sample file without FX running it would be more helpful to understand your situation. Something with just 5 seconds of background noise followed by 10 seconds or so of a couple phrases would be ideal. If you are not comfortable sharing it publicly, you could send me a PM and I can take a look at it. I have been working with someone using the AT2020 for narration, and a chunk of the issue was resolved with sound isolation, mic rotation, and how the mic was being used. A HPF before the Sonitus took care of a good deal of the hum and let the gate operate more smoothly. Specific to the Sonitus Gate, the threshold (input slider) shouldn't be too far above the noise level (and as far below the signal as is tolerable) during mic checks. The "Depth" then needs to be adjusted to prevent the chatter (it defaults to -inf, so will slam shut that way... her situation was around -25dB). The other default settings were close, but I think the Release got bumped up to around 400ms. That said, every recording situation is unique, but the HPF, Input threshold, and Depth are where I would start.

Back to @jesse g's question, I am pretty sure that cross grade has dropped to roughly $183 on JRR before during the Holidays, but now I am wondering if that was specific to the Gibson debacle. That event actually happened right before BF, and a lot of vendors went out of their way to accommodate all of the SONAR refugees that year. There is often a lot of hoopla with the .5 releases which confuses people into thinking they are another purchase, but they are included in the main versions. I never realized there even was a Dolby Atmos VST until I saw it in here on sale for $179... PreSonus gave that to us with SOP 6.5 for free. All the hype with the .5 releases may be to get new folks to onboard during the off years, but those releases tend to have a lot in them.

Hmmm, not sure what to make of that once I read it. It comes across as a VSTi being inbound and a quick way to build a sample base for it, but the dry/wet runs hint at a profiler being inbound as well.

The Holiday season tends to be the best deals. There is often another one (often cheaper) done before the next version release. With current timing, that one would probably be next summer (and also outside the window to get a free upgrade to V7) when it hits. I would expect V7 to hit Sep/Oct of next year or thereabouts. I just picked up the Revelator condenser for $59 during the second Amazon Prime Day, and the one-time software option for that was either 1) Studio One Artist (outright), or 2) Studio One + free for 6-months. While the Studio One+ is an option, it is also an (expiring) subscription, which is roughly $140/year. By comparison, Studio One Pro (once you own it) is roughly $60/year to maintain it current (typically a sale at roughly $125 or less on JRR, with the version releases every 2 years).

I totally forget when this was implemented now, but Cakewalk used to use a Global Audio folder (what a mess that was), and you could actually carry that setting forward if you chose. If there is an audio file you think is "missing," yet it plays, right click on it and choose "Associated Audio File..." at the bottom. The "Path" that it pops up could very well be the Global Audio directory, which will match the Preferences->Files->Audio Data setting (defaults to "Cakewalk Projects\Audio Data"). Also, there is nothing preventing a cwp from referencing an audio file anywhere on your machine (even from a previous cwp in another project folder), so the Associated Audio File... will let you see what directory that file is really in.

It basically boils down to any track-based operations (like resizing, etc.), many of which can be done on multiple tracks simultaneously, require the track number(s) highlighted (selected) so that CbB knows the target of the operation you are performing. As with many programs, a lot of user errors come from operations without explicitly selecting the objects (to be acted on) first.

Thomann specifically mentions SpectraLayers Pro (but not the version) with the Samplitude Suite above, but not with the Sound Forge Pro. It also doesn't mention which Sound Forge version is in the Suite (I "assume" it is 16), so something to be aware of... it is very possible that they are the same and just data entry errors, but cannot tell from just reading them.

Historically, this has been one of the cheapest methods to onboard into SpectraLayers Pro for a new user (especially with the Samplitude Suite). Some of the other applications (Melodyne Essential, Convology XT Complete, etc.) also have a fairly hefty "list" price as well. For someone new to all of these it is pretty substantial, but loses value if you already own the piece parts or an upgrade to them. This is the first time I noticed they gave Samplitude a face lift! Anyone know offhand when that happened? (Last time I got that Suite it was X5 and still looked like a Win95 program).

Something to bear in mind is that even with 24-bit ADCs, there are enough interfaces out there with pre-processing FX that nothing would prevent them from passing 32-bits to the host. Implying that they are 32-bit ADCs is bit much, but pre-processing is essentially doing the same function as the DAW, and no reason they cannot pass that on.

Poor OP is getting inundated with the perpetual discussion/debate, when this was all Max was asking. The practical side of this (beyond carrying precision through calculations) is that a signal cannot be digitally clipped inside the DAW. The downside is that a rogue plugin can spike pretty deep into the red on you, which is why a limiter on the Master bus is always prudent to protect your hearing. ***** This is OFF TOPIC and more for John: I was testing out the PreSonus Revelator (condenser version), and got a pretty pronounced latency at one point (more annoying, but usable) from inadvertantly having the mic and CbB at different bit rates. I chuckled when I found that and thought that has to be one of the more impressive things I have seen in a long while... that mic was working like crazy to "trick" everything it was connected to to think it was working in their native bit rates. Even the CbB preferences window took like 30 seconds to open (with everything still connected), but PreSonus had no issues playing the game. DISCLAIMER here... I would not recommend the condenser version to someone new (especially for recording voice with environmental noise and sibilants concerns), but have not gotten to test the dynamic version (yet). That mic is actually its own audio interface with its own ASIO (and OBS) drivers and counts as Dolby Atmos hardware.

I just took a quick look at the manual and AFAICT the USB is only passing MIDI data. With the Drivers installed (and the Korg connected), you just need to go into CbB preferences->MIDI->Devices and check the box as a MIDI input (similar to what you did using your interface). Check and see if it is listed in your MIDI devices when plugged in via USB. You probably can also check it as an output, but to get audio from the Korg back into CbB you will need to run audio from the output jacks on the Korg to your audio interface. In fact, you can record into CbB for both MIDI (via USB) and audio (via the audio outputs to your interface) but may see some latency in doing so.

Quick follow up with the OP on the BL Denoiser specifically. I have never been a fan of processing things you really want to bake each time the transport runs, and the BL Denoiser loses its settings in addition to causing latency. Because of losing its settings (which also causes it to phase), this plugin is best suited for a "once and done" workflow. After insertion it does extremely well, it just cannot be left active. The quickest method I found for using it (almost identical to a wav editor actually), is to insert it as a clip FX (not track), loop an area of ambient noise and toggle the Learn on and off with with the transport looping, adjust to taste and then Bounce to Clip(s) (which bakes in the clip FX). BL Denoiser is actually very effective when used in this manner.

Thanks for posting this. Just the description in that alone is awesome! "PersistentWindows helps rectify a well-known and frustrating multi-display bug. This bug changes window positions and size when the monitor display count/resolution adjusts and restores its previous settings."