mettelus

Quick question to clarify the OP... are all of the tracks synced to each other? One of the quickest (manual) methods is to Set Measure/Beat at Now (Shift-M) as you work through the track. As long as the tracks are in sync with each other, this will let you insert a tempo map point each time it drifts. Typically every bar or every other bar is common, but sometimes you need to go to a beat level if it is all over the place. This does not adjust the audio at all, it only adjust the tempo map to line up that audio.

I still get a chuckle from the age reduction CGI used in the new Stones release. A millennial going to see them in concert might come out in shock and say, "I think I just saw 'Living Dead - The Musical."'

I suddenly realized I posted a video for using Adobe Audition in SONAR almost 10 years ago now. I have no clue what I used for the video capture (use Camtasia now), but even the cursor latency in "whatever" I used sucked. I actually have "Audacity" in that Utilities menu, but never used it (that I can recall). The only difference between then and now is that I normalize to -3dB by default. I remember someone had asked how to do this, so threw this together to demonstrate it way back when. Where I "cheat" with this (and do it often), is that non-optimal capture environments (with steady state noise) are no big deal as long as I get a clean capture of that environment in the initial recording as well.... always a pre- and post-roll capture on things. I did one once about 10 feet from a fireplace starting to take off (so the post-roll was louder), and Audition removed that without issues... in fact, the one in this video may be it now that I look at it more closely.

Audacity can be added to the Utilities menu just like any other program, but if you can find AA 3 (there was a great Adobe server debacle years ago on the old forums where folks could grab Creative Suite 2, which had AA 3 in it). I use AA 4 (from CS 5.5), and IIRC the noise removal algorithm is identical between AA 3 and AA 4. That noise removal is far superior and highly recommended if you can reinstall it. I do not know of any DAWs offhand with this feature but there may be... it seems to be more a focus to wav editors or post-production suites. The AA 3/4 version remains my preference just due to its speed.

I understand where you come from and see that sentiment often. I come from a background where signal-to-noise ratio reigns supreme (to the level of being alive or dead). Signal = source level - propagation loss, and Noise = environmental level - directivity index (the ability to discern that signal discretely even when buried). Signal is all that matters with processing, so obliterating noise completely allows for those processes to occur readily and without limitations. I did, however, run into someone who has a -60dB noise floor requirement recently (for narration). After that "GTFO" moment for me, I thought, "Okay, I'll play," and went about business as usual, then introduced them to iZotope's Vinyl. If you use only the mechanical noise module, the date buttons at the top control a noise band that gets lower in frequency as you go from 1930-2000. End result was to use the 1950 setting at -19dB to insert that "-60dB noise floor" on the master in a region that was easily drowned out by the vocal fundamentals. I still chuckle about that "requirement", but whatever.

+1, I am firmly entrenched in using an old copy of Adobe Audition in the Utilities menu so I can do that in less than 30 seconds and not require any track bounce afterwards, but I am also always on the lookout for options to new folks that function and don't require massive money investment. Spending CPU power every time the transport runs to remove something that you want to obliterate anyway is a waste of computer resources. I was hoping this might have tracking potential, but the overhead is a bit much to be feasible.

Quick follow up on this guy. In practice, running this enabled can cause some latency issues, so I am not sure about the feasibility of running this in a tracking situation. I got it to cause its own phasing issues, so be conscious of the algorithms used and how much tweaking you are doing. It definitely raises issues with plugins that have look ahead embedded in them, so its best use is as first pass for post production, then bake that into a track bounce so it can be removed from the process chain completely.

I wanted to do a quick post on this to close the loop in case anyone comes across it later on. The condenser version of this is extremely sensitive, so I did a deep dive into this to find out if it was electrical or mechanical. It is 100% mechanical. The desk stand that comes with this mic is its worst enemy (very rigid) and the boom mount attaches to the bottom (which is just funny to look at), but I attached it to a boom and then folded a towel to over 1" thick under it (I have hardwood floors), and the problem disappeared. Totally functional, and with ASIO drivers used it can be driven at very low latencies (buffer goes down to 16, but I was running it at 32 with a full FX chain enabled internally after I ran the noise issue down). The dynamic version shouldn't have the same issue to such an extent (that wasn't on sale), but for the condenser version some shock mounting is required (the included stand is a piece of meat).

Even in Windows 10 that happens a lot when connecting different devices or altering their settings; always start with Windows settings, then drill into apps. I have recently found out that it is still as big a PITA as it was to change "sound profiles." When starting CbB without an ASIO device connected (when one was previously used), it won't just shift modes to a previous (available) setup. I have to go in and manually and change driver mode, then select a device, then wait for the profiler to run. I don't have any other programs that do this... they just say "this wasn't found, want to use that instead?"

For reverbs especially and time-based FX in general, you want them post-fader (the default) unless you are looking for something unique. The reason is that you want the FX to be proportional to the actual track signal so it sounds natural (those FX should never be stronger than the source in real life). If shutting post-fader off "solved" the problem, check the routing on that track to understand what is going on. Post-fader send to a reverb bus is expected for typical use, and there is very little between a pre- and post-fader send (maybe a panning mismatch? or the fader has a hefty gain increase on the track).

I am confused by this as well. The auto-gain feature is pretty much identical to "riding the fader," so it only needs adequate look ahead to hit the target you specify. Maybe there is confusion about it fluctuating over time (which is how it works)... it is NOT blindly applying the "same gain" to the entire track (that would be no different than Process->Apply Effect...->Normalize in CbB which is instantaneous). That plugin is constantly adjusting the incoming audio to hit the target you set.

Be sure the sample rate in Windows itself also matches for the device under Sound Options and that it doesn't give exclusive control to apps. Essentially Windows can lock sample rate on you too. I typically go a step further and make sure Audio Interfaces are not set as default Windows devices, just to keep Windows' mitts off it. As far as changing sample rates. Offloading stems from a project and then batch importing them into a new project (set to the new sample rate) will SRC them all on import.

Do you know the motherboard in the machine? I would start with the manual for that motherboard and look at what it is capable of handling. Reason I say this is that you might not be limited (and probably are not) to a PCI Express 2.0 for your options. I had to check what I did with mine, but replaced the GTX 580 (PCIe 2.0) I had with an ASUS STRIX GTX 970 (PCIe 3.0). Once you know what the MOBO can handle, https://www.videocardbenchmark.net/ is a good site to check performance of various cards you run across. Bear in mind that NVidia owns the architecture, not the manufacture, so even cards with the same number may have differences in them. One of the GTX 970 brands was designed like a jet engine for cooling, so was a poor choice for a DAW machine (why I went with the STRIX design).

Just got to check this out, and being a plugin this has another advantage over editing suites in that you can use it and leave it enabled while tracking. Although it will still embed that noise into the recorded audio, it alleviates all of the distraction issues while tracking. This is definitely something worth having in one's toolbox. The 64-bit version from the OP has a much better resolution in the spectrogram than in that video teaser. This is also why I harp to some folks on pre/post rolls while tracking with non-optimal setups... without that background noise totally exposed, it is incredibly difficult to address in post production.

I didn't even know this existed. A lot of folks have asked about free noise capture/noise removal methods in that past and this one totally fits the bill. Thanks for posting this.

Correct, the auto-insert of TTS-1 will only occur with no MIDI outputs selected. Although it is not a huge deal to add external synths once CbB is opened, it can be a PITA to remember to disable them when closing CbB. As far as routing, I would highly recommend considering track templates for any instruments you take the time to set up. After loading a MIDI file you can simply insert track templates and route the MIDI outputs as appropriate, OR... another option is to simply shift-drag MIDI clips into the correct synth track from the open MIDI file (even with outputs selected on load), which may be far quicker. If you get to a complex project that you use often, a project template would be helpful, but there wouldn't be a way around getting MIDI data from the MIDI file itself to point to the correct track other than mentioned above. There would always be some "sorting" involved.

The oxidation layer that forms on electronic connections is typically very thin, but it can cause issues over time even for things that are permanently connected. That layer tends to be so thin that simply reseating/reconnecting it will usually wear enough off enough to re-establish a good connection. Connector cleaners like DeOxit are also something to consider for cleaning those (whenever you happen to open them), as well as for things that are important, but not always seated (XLR connectors, wall outlets, and the like). Those oxidize quicker because they are open/exposed more often, but those are also worn down each time you make the connection through simple use.

This ended up being very good for its purpose. The only downside to it is that the noise floor is a bit higher than anticipated (most reviews reference this), but the internal gate was able to remove that with a little tweaking. Being the condenser version, that gate (and other internal FX) are required to bypass external processing completely. The only other nuisance for me is it comes with a 3' cord, but I used a 10' cord and it didn't care. The latency is enough that I wouldn't use this for musical tracking, per se, but it definitely hits the mark for screen caps and podcast type work. Upsides to this guy is that it does have a pretty complete FX chain built in. Although the order is locked, if you go into the advanced setup some of the modules can be swapped with other emulations. It comes with ASIO drivers, which includes 2 virtual inputs as well as one for the system connected to (4 total when you include the mic itself). Three polar patterns, three output mixes available, OBS capable (PreSonus has their own OBS installer included), and ironically (for me), this is also Dolby Atmos capable, so that binaural demo they did for Studio One 6.5 I could actually listen to. So far I have only used it for a couple screen caps, but the virtual routing has been a welcome change for me since the DAW output was now crystal clear. All inputs/outputs are exposed in the DAW, so I could send the Master bus to the mic's internal mixer, then one of the mixes to the capture app. Everything is exposed, so I could have done a loopback into the DAW as well, but the internal FX needed no post processing so was moot.

One common chorus effect used over the years is to duplicate a track, pan one hard left, the other hard right, then scoot one of them by roughly 30ms (plus or minus). Fiddling with that delay time will provide varying effects.

Depending on the age and type of mouse, disconnecting and reseating connections may help. If it is an optical mouse, also check and clean the optics socket. The oddest mouse behavior I ever had was from a piece of lint in the optics socket.

Don't feel bad... all of the music that comes from my other end always becomes sewage as well.

Or... you could get the pre-owned one for $124.65!! This deal is also on Amazon (one day also for the Prime deals). On there it also has Ableton Live Lite listed with purchase. Since I do most screen captures with a webcam mic, I am actually interested in the pre-mixing features this has to see how it does.

Google Christian Henson, that incident is pretty high on the list of hits

So it was YOU! LOL Over the years I have become a fan of gallon freezer bags. They are big enough to hold "sets" of things and I can just label them with a sharpie and throw them in a box without having to dig too much later on. They also keep things from getting dirty or dirtying other things (the freezer bags I consume the most are actually kept in the garage so I don't lose tiny parts or transfer grease to everything).

There was a site posted a while back to give you insight into the frequency response of various mics using the same audio. I forget where it was offhand, but the real takeaway is that frequency response is sort of moot with proper EQ (what that site was actually doing was overlaying the mic-specific response curves over the audio samples). Adequate signal-to-noise ratio tends to be more important so that you can isolate the signal and tailor it accordingly. As long as the frequency content was captured, one can process it at will.