mettelus

Out of curiosity, are the samples with BFD3 wav files or in a proprietary format?

+1 on recording synths to audio. Another simple hack is to mix down the entire project to a single track (pre-master). You can then simply drag/drop that into a mastering session or folder of your choice.

+1 on Andy's comments. Either a large diaphragm condenser mic or good dynamic mic through an audio interface is ideal. The Blue Yeti is locked to 16-bit, 48KHz, so another thing to try is make sure the DAW project reflects that before recording, and probably WASAPI shared for drivers settings. You may be seeing mismatch in bit depth/sample rate, but even if this improves you are not going to do your audio recordings proper justice. Andy's recommendations are spot on.

There is some validity to this comment, and let me explain context. Until about 5 years ago, and old friend was still using a 4-track to record, and what was on those was pure tracking performance (vast majority was one take for guitar/vocals, drums and bass were programmed into a machine). During the discussion, his advice to me was "Don't track anything until you have the composition fleshed out." My advice back to him regarding a DAW was "Don't get caught up in an editing loop for something you can re-track 100 times faster." He already thought that way, but I cautioned him not to lose that skill. Using a DAW for composition is almost required for instruments that go through a DAW anyway (i.e. the instrument is a VSTi), but I have noticed a good break point for myself is to stop when to a demo mode, take a break, then start a new project for the final version. A lot can happen during that break, and starting a new project puts the focus on performance rather than editing (but I do bring in tracks from the old project that I am not going to perform... drums and bass almost 100%, same as him). Ironically, when I got Melodyne he sent me a 7+ minute master and wanted me to change one note. I laughed and told him 1) Melodyne is not a miracle worker and 2) sometimes performance and composition differ (which is good)... the only person who doesn't see that as a perfectly acceptable accidental is you 😀

I admit I am confused about the width comment for the Ebow, as I have used one for 30 years and always on humbuckers. I do not see split coil referenced in that guitar's specs. An alternative, is to add a split coil feature to the installed humbucker(s) (but not sure if all 4 coil wires are exposed). I have 7 switches on my main guitar, and 4 of them are "hidden" in Seymour Duncan Triple Shot Mounting Rings (configured to if switches apart it is parallel HB, together is series HB (standard), and if in the same direction for the single coil they are pointed to). This setup also allows "humbucker options" by selecting a single coil in each pickup (if both are done), removing the hum but changing the tonal quality depending on spacing from the bridge. Visually, they are simply mounting rings, and all wiring is in the ring itself, so no guitar mods (even to wiring) is necessary. The rings output go to where the humbuckers were.

Loopback is the feature you are looking for, and I do not see it referenced in the AudioBox USB 96 manual. Some interfaces have the feature internal to the device, but not all have safety nets in place to prevent a positive feedback loop occurring, which can damage the unit, speakers, headphones, or worse your ears. Warnings with this setup, especially if you are unfamiliar with the unit: Always put a LIMITER on the master buss output to the audio interface just in case. Because not all limiters are "brickwall" limiters, set it to -6dB to ensure it is caught before the interface can see it. Protect your equipment and hearing. You can always adjust this once you get it working properly. If the input being looped back is a track inside the DAW, muting that track will still make it heard in the mix, but prevent it from being compounded into a feedback loop. Work with low volumes (especially the headphone/monitor outputs) on the audio interface until you know the signal path is correct. If the master buss suddenly pegs with input, do not hesitate to kill the audio engine in the DAW. The above is especially true if you ever use a "physical loopback," where you run patch cables from an interface's outputs back into a set of inputs on the same unit.

Plus he soon learned that babes flocked to him and it increased his social status. David Former Barbarian is now David the Gigolo... the rest is a musical soap opera.

Hmmm, since computers (and especially DAWs) didn't really exist when most of us started, maybe "Blow off all of this expensive hardware and focus on performance... personal computers will be a thing soon (seriously), and you can get all of this gear for a fraction of the price."

Files cannot be directly attached to forum posts. The file would need to be uploaded to a hosting site (One Drive, Google Drive, etc.) and then the link can be posted here if you want to share it. Unfortunately, some of the software best suited for what you want do does not have visual assistance, but there are a lot of folks in this forum that have them and can help you out. Just an aside if you want to post a link, if you can specify what section you want (bar, 2 bars, etc) it will help us out if it is not obvious. Also, if you do not want to post it publicly (forum posts are permanent record), you can send me the link in a PM.

Quick note on the above, I still use an older Saffire and File->"Restore Factory Defaults" in MixControl does not overwrite settings saved to the hardware. In fact, the workaround for the v3.7 bug from way back when is to do that, and then File->"Load From Hardware" to properly initiate the unit for me. Once you have Mixes and Routings set up to your liking, saving your preferred "master copy of settings" to hardware (in addition to your PC) is convenient. I missed this thread on its first pass, but the OP may have been able to restore from the HW if it had ever been saved to it.

For the MOTU, you would need to set that as the default Windows device in Sound Settings->Sound Control Panel. Then right click the MOTU->Properties. On the "Advanced" tab at the top of that, uncheck "Allow applications to take exclusive control of this device." The hurdle with doing that is that the first application to access the MOTU is going to lock its sample rate, so you are going to need to use one recognized by Zoom (if applicable), and not have other software running that would access it on you.

Ramping up volume is similar to turning up a stereo as a songs fades. At a certain point the noise floor is your enemy and becomes audible, then overwhelming. Without seeing the clip specifically, that is worth a try. I have never owned 8.5 but realize why you still use it.

Granted, iZotope is very good about cleansing older copies/references of its software, but RX 6 (2017) was not the first version. https://www.soundonsound.com/reviews/izotope-rx https://www.izotope.com/en/company/press/2010/izotope-releases-rx-2-complete-audio-repair-suite.html

That is true, threads tend to get dredged up after a few years and your list of audio interface questions is timeless. I hadn't looked at the Solo since it came out, so was pleasantly surprised to see it got some real upgrades. When first released it had RCA jacks on the back and now they are balanced outputs. My FW card (not the Saffire) is starting to degrade on me so been debating just shifting to a USB interface since I could actually carry it someplace else and use USB 😃. I was surprised Amazon has like 48,000 reviews on the Scarletts... that number was way higher than I expected.

I was wondering this when I updated it last night too, I guess for PC users it didn't really have much effect other than to change the version number for us.

I couldn't remember what Ableton used to cost, so put "Ableton Live 11 Suite" into Bing and clicked the shopping tab. First two hits were a site with it listed for $70, so I checked to see if that is a legit site (supposedly is), but left me wondering how they got the serial numbers. Ironically, Bing doesn't point you to any reseller you would expect to see, so I didn't get my question answered

I hope the OP doesn't get overwhelmed reading this. We jumped in a bit deeper than he was anticipating already.

It also depends on computer resources and which Melodyne variant you are running in a way. I do not leave Region FX enabled, so would bounce them to one clip, do one edit session and bounce out of Melodyne to save time and get it in one pass.

The above will get rid of the excess data. To find the offending track zoom out in Track View and one (possibly more) of the tracks will have clips all the way to bar 100.

I checked quick on that UM2 unit just to be sure. Sweetwater has a comment on setup "Although most Behringer audio interfaces have their own driver (version 4.38 as of July 2018), some of the smaller interfaces will require the third-party driver, ASIO4All. These interfaces include the UMC22, UM2, UCA222, and UCA202," so it is an ASIO4ALL unit. I also didn't realize that was a 16-bit/48KHz unit, so if possible to return that, it would be something for the OP to consider. ASIO4ALL is really a WDM wrapper and tends to be problematic, but in certain cases is usable (I have used it for playback of stand-alone instruments that require ASIO through the PC sound card). There have been so many posters over the years seeking help on getting ASIO4ALL to work properly for recording. Side comment: I very much appreciate the feedback forum members over the years! When delving into unknown territory that has been a tremendous help, so thank you for taking the time to advise. The "trick" is asking before making big leaps.

That search engine needs to be updated. That DAW column is not accurate at all (he is promoting PT), but people do need to know plugin format. Quick glance there are a lot of VST plugins he has listed as "PT." Being able to filter out AAX would be useful though.

Thanks for the confirmation on the M2. I thought that unit was in the "ASIO4ALL" bucket but wasn't sure. Back to the OP, I am pretty sure there are a fair number of forum members who have learned the hard way (including myself) that skimping on an audio interface is only going to buy you a fair amount of pain and suffering for no reason. If you want to get audio into a computer and use a DAW to its potential, an adequate audio interface is a must. It is not like we are trying to sell you something you do not need, but rather offering lessons learned on what not to do. DAWs, instruments, FX, and the like can easily fall into the bucket of personal preference, but a good audio interface is a universal piece of gear that is required to get audio into the DAW. There are many companies with liberal return policies, and one of those is highly recommended to test "unknown" gear. Once you get one that will let you see the night vs day difference in what you are trying to do, you will end up kicking yourself for the time wasted getting there.

Since you have that unit already, I would recommend searching specifically for questions related to issues you are having. I am not familiar with that unit, so not sure if it has dedicated ASIO drivers that come with it (this is sometimes the most critical). As far as setting things up inside CbB, SWA did a very nice series of videos on SONAR X2 years ago. A lot of that is still applicable and the first three videos cover audio/MIDI setups (the chapter listing for that video set is here - SWA posted 50 videos on YT without an index!).

A dedicated audio interface is what is required. Most of the budget models are going to run around $100, and it is not a piece of gear you want to avoid getting if you are intent on getting external audio into a computer. If just recording yourself, the Focusrite Scarlett Solo would be a good option (1 microphone input/1 instrument input). There are other options in that price range, but Focusrite tends to have fewer issues noted with them overall.

The above responses cover your other questions well, so my response if more for this one: This question often comes down to if a synth is being played solo (or as the primary instrument) versus being used in a mix. When solo (and synth presets often reflect this), they often cover a wide frequency range with a lot of FX to draw attention. However, if using a synth in a mix, this is something to be wary of. When mixing, EQ (not the only method) is used often to resolve frequency conflicts between tracks, and if "too processed" this can become an issue. Specifically, time-based FX (delay, chorus, reverb, etc.) are often used last in FX chains because they make "multiple copies" of the source material you are trying to mix into the final product. This is not a rule (as with most things musical), but when highly processed synths are involved (especially ones covering a wide frequency range), those will get a lot of attention during mixing to ensure they do not have frequency conflicts with the other tracks, which is easier when time-based FX are not yet present (often added after this step). Concise answer is that, when mixing, it is often best to apply time-based FX to tracks/busses after they have been assembled with other tracks to fit into the mix properly first.