mettelus

Turning off monitors will rarely affect things, so that is typically fine to do. However, both sleep and hibernate modes have issues native to Windows (not the DAW), so those are two you should avoid. Windows has never been elegant with sleep/hibernate modes, and has internal functionality that is hard to defeat or even figure out at times (like shutting off hardware/ports that do not come back online as they would from a boot).

Windows will only see devices directly connected to it as stated, so as a pass-thru only the MIDI port on the interface will register. You can lose some of the internal functions of the keyboard doing this, but it will pass MIDI no problem. 5-pin DIN connnectors into the computer are rare anymore, so many use 5-pin ports being available as a criteria on interface purchases if they need them.

Quick edit: I am not familiar with that keyboard, but if you do have USB available from it, that is highly preferred, since it will make the keyboard "fully functional" with its own drivers. The USB connection will pass everything the keyboard is capable of doing, versus the pass-thru route.

You do want things connected (to the same ports) and powered on before opening CbB by default. The simple reason for this is that CbB reaches out to Windows to see if your CbB preferences match what Windows shows available. Once you have everything online and setup in preferences (what you may need to do when/if you have connected things CbB has never seen before), if they are always available when CbB launches, it will "just happen." Even silly things like swapping USB ports will make Windows see it as a new device, and this can trip up CbB because it asks "What is available, and is it known to me?"

When you said, "I get a clicking sound then there's no audio at all," is the audio engine stopped (and can it be re-enabled), or are you sending MIDI to the keyboard? This may simply be a preferences/routing adjustment that you need to do once to correct, but after that the project should come online without issues so long as devices are online before the CbB "looks for them."

1 hour ago, norfolkmastering said:

Can I ask what you mean by 'in the Media Browser'?  Do you mean where the .cwp file is stored in Windows?

And when you drag the second project to current project's TV, do you mean into the Cakewalk GUI?  Where exactly do I have to drag it to?

Thanks

The B button is the default shortcut to toggle the Browser internal to CbB open and closed (is docked on the right by default).  The "Media" tab in the upper left is similar in functionality to Windows Explorer, and if you click the drop down arrow, there is probably already a "Project Files" in that list that goes to your top-level Projects folder. When you drill into folders, you can see the cwp files, and can simply drag/drop one onto the left side of the Track View to load it into an existing project (bit rates must match though).

Welp, it just took a little more digging.... AA3 DOES work, can be installed, and is so old (2008) that it has a manual registration cycle (I simply said "Do not register" when I got there). The "correct answer" in post was accurate. There is a "components.msi" file in both the 3.0 installer and the 3.0.1 patch that worked for installing the application and patch (the setup.exe files both fail on Win10). Upon launch it failed to load twice because of incompatabilty issues, but the Win10 trouble shooter popped up and fixed that. The layout is different from what I am used to (with AA4), but the noise algorothm seems identical (the only difference in a null test was default differences between the two apps on the default render tweaks (I am not in the mood to run that to ground and no one would know/care anyway)).

I was bothered with the initial pass I tried above because the default installer STILL downloads all of the needed files from Adobe's server (just the setup.exe won't run in Win10). Even though I saved that download folder from years ago just in case, it bothered me that the installer was still able to pull them down without an issue but not run them.

Quick FYI, I guess Adobe's reaction to the debacle a few years back was to shut off the activation servers completely for CS2, so AA3 won't activate even if you can get it to install. I "think" that was scripted for XP SP3, but even in compatibility mode it won't go past the choose language screen on Win 10. The activation servers being shut down makes it a waste of effort anyway.

I guess I need clarification on this still. Your first reply made me think it was a simple "tempo map doesn't follow the performance," but when you said

7 hours ago, Michael Fogarty said:

I have to cut and move his guitar playing and voice many times through the song

It sounds like you are doing a multi-session project and what he has done so far is synced to the tracks he performed, but not to the tempo map, so that you can edit with snap-to-grid enabled and line up additional tracks. If you are doing track-based surgery, ripple editing seems to be what you are looking for (i.e., not aligning the tempo map to his performance, but altering his timing to make it fit more appropriately). Creating a tempo map will not alter the audio, but once you say "cut and move," that sounds like ripple editing (perhaps on a track-level, and not a session level). Is that accurate?

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.

6 hours ago, Keni said:

Remove the noise and get rid of it!

+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.

14 minutes ago, tiyaf said:

I have an old pc which works great. Currently I have a Nvidia GTX 570 PCI Express 2.0 card. I want to upgrade the GPU. Any suggestion for. PCI E 2.0 based latest cards Thanks

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.

On 9/27/2023 at 8:37 AM, rfssongs said:

checked a few connections, reseated the memory

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.