mettelus

They are free and do not take up a lot of space. The disadvantage is that some have FX baked into them (either negating raw samples from being used or requiring a bit more post-processing to manipulate them), but "as is" most are creatively designed to give them a unique flavor unto themselves.

iZotope's Vinyl has been around a long time. Be sure to remember you have it if used in projects. It is one of those plugins that inserts noise with the transport stopped. Remembering you have used it will save you some heartache trying to troubleshoot your audio interface in the future!

iZotope's Vinyl is free, and I think that has a decade dial on it but forget for sure. Check a recording that you really like for frequency response. The high end might be rolled off more than you may be anticipating.

This is Motherboard dependent, so be sure to check this information as you build your machine. Typically one of them can be set to higher speeds (X2 or X4) at the expense of SATA III connections on the board (my NVMe X4 drive disables SATA 5/6 connections). I used normal speed for the OS, and the X4 for a larger sample drive. Otherwise, your intended configuration will work, but make sure the motherboard supports the NVMe drives too. Another thing to check is the form factor and keying for NVMe.M2 drives... one is for their length.... make sure what you want to use will also physically fit onto the motherboard.

Filters also attenuate the desire signal, so noise reduction that can capture a noise print (part of the file that has nothing but noise in it) and then removing that from the entire file (as long as it is consistent noise and not transient-based) is the most effective. The two best option for this are Audition and RX, both of which are not free. Audactity and ReaFIR are both free, but do not do quite the same thing. Is there a portion of that file that does not have bass notes in it? Is that hum consistent in the file? If yes to both and you are willing to post the file, someone with Audition or RX can address it (if you can post the link here). If desired, you can also send me a PM with a link and I can do it after the sun goes down today. As an aside, the Capture Noise Print/Noise Reduction is an effective tool for any audio recording, but requires a portion of the file to be "just ambient/environmental." For that reason, I will record lead in/out portions of an audio file if I have environmental noise. As long as you have the tools that can remove it properly, it saves a lot of time in catching ideas on the fly (particularly microphones) and recover them to a condition that can be subsequently used. I kept a file from years ago that had -40 dB of fireplace noise in it that got slightly louder over the 90 second recording (I do not noise gate above -55 dB in most cases). Audition removed that so impressively that my workflow has relied on this since. This has also been an ongoing Feature Request for years to have included in SONAR/CbB.

The only reason that I can think of for boot disc "size" is for imaging the OS, which is the only image I perform. The smaller the image, the faster it can be done/restored. That being said, with NVMe drives, most machines are limited to 2 these days, so maximizing capacity for those is more cost effective. From an image standpoint, that is the only situation I would partition a drive these days. But in doing so, also realize that a physical failure of the drive can lose it all, so backup the data on the secondary as well. Consider the imaging perspective when choosing your OS drive/partition. I image the OS, but use xcopy/robocopy on data drives since it is far quicker. For an OS that small (mine is 256GB), bear in mind that I also riddle my OS drive with junctions. I let software install to default locations (except VST dlls), then end up moving/junctioning anything that gets larger than 4GB. My OS drive is 256GB, and runs around 130-140GB and images less than 100GB. When keeping an OS that small, you also want to be aware of how to clean the rubbish off of it (temp files, downloads, etc.), especially before each image.

+1 to this. The only issue I know of is backwards compatibility with Aux tracks/Patch points. Forward compatibility is not an issue (X1 files open in CbB). You will simply get a warning that "x VST failed to load" without re-installing those missing VSTs, but all MIDI data and audio will be there. Chuck's cleanup advice is definitely advised. Anything you want to pull into SO as a "project," I would also recommend exporting audio as broadcast wav files, one per track, so all audio will line up for you during the import into SO.

Be VERY careful doing physical loopback, they can create feedback loops so quickly that they can damage equipment. Do NOT just attempt this. CbB will record to an armed and muted track which is "typically" the safest way to prevent feedback loops, you do not want to reintroduce output back into inputs. Basically you want monitoring and inputs separate, which needs to be understood before trying this. Also, always start with gains at zero on inputs before introducing signals or making/breaking connections. Before you take off attempting this, understand the above and the details of your gear. I am on a cell now, but wanted to inject some reality before you get into issues that could destroy something. Most physical loopbacks (if not internal) have NO safety features to prevent damage to components.

The OP references not being able to open files because the sample rate is higher than the sound card. I have never heard of this, since CbB automatically streams to the current sound output capacity. You "should" be seeing an error on open with the option to select another output source, as simple as that. That file not opening could be internal corruption or another error (there have been posts similar). I would recommend contacting customer support if you are not seeing the option to choose another output device. Azslow3 created a plugin for Reaper which converts cwp files to Reaper format. If that cwp file will not open, that is an option to recover the MIDI information. Numerous free wav editors will convert the audio, which should reside in the project->audio subfolder.

I didn't even realize this existed and was a little shocked that it is a paid add-on. Every other program that I know of for batch conversion has it baked in. SO lets you drill into the wav folder by opening Windows Explorer, so I am a bit at a loss for the excitement for it. Even the script processing has been out for years (at least for Adobe Audition 4+), but that particular feature is very nice for dialog work or instrument sampling, and not sure if that exists in anything that is free. The price point seems to be a bit high to me for simply being integrated, but the scripting feature may be the real reason for it.

TTS-1 does have the advantage that CbB (with no output MIDI selected in preferences) will launch MIDI files, insert TTS-1, and route it for you. That feature is nice if you are starting with another's MIDI file, then can replace content (shift-drag MIDI) to other VSTis. I'll admit that with the processing power of newer machines, I do not get as excited by multi-tibral as I used to. For some reason I waste more time routing/setting them up than to just pick and choose what works best piecemeal. Then again, it could just be laziness on my part.

Another comment to add to the above. The SI bass is also off by an octave IIRC, requiring you to to manually offset it. Unless you also have notes in the higher octaves, it will appear to be playing, but no sound.

I just ran it again and it came up fine with the Install option for the update. Launched fine afterwards as well (no need to refresh), so I think they were fiddling with things on the back end while this thread has been going.

LOL... this is why I don't pounce on updates too often. My app is just saying, "Something went wrong. The JSON response was invalid." The only option is to simply exit the app. I will hold off on this till the next monthly freebie drops.

This is a plug for MCharacter from last year (old forum even). For folks who are on the constant search for "new samples," MCharacter can do a lot of incredible things even with mediocre samples. It works best on monophonic material before an FX chain, but has creative uses as well. Well worth running the trial on it to check it out, and I am sure will be back around on Black Friday at 50% off again.

Realtek has released an ASIO driver, but I have never seen it function, so would recommend ignoring that one. The Realtek is adequate for audio playback, but not audio recording. I am pretty sure that is still 16-bit only, so you could also be seeing bit depth/bit rate issues trying to record with it (the options it has internally via Windows are pretty limited). After updates, Windows could potentially assign the default sound device to something that doesn't actually work (like an HDMI monitor), so start with Windows sound settings (in 1903 there is an extra step getting into that via a link in the upper right of the sound page).

I am not sure the correlation here. Given the track record of OSX updates forcing developers to update (or everything spontaneously breaks) comes across to me more as foreshadowing a possible, and potentially probable, future. There may come a time when the overhead to keep things operational isn't worth the effort to do so. The lead in to his article is to the point "Over the years, many users of non-trivial applications have reported compatibility problems in OSX and it looks like there will be even more reports in the future, starting with Catalina. This may be a good time to review your options and consider switching to a different platform." Again, I am having a hard time seeing how this comment is taking anything out on anyone. This comes across more as stating a valid risk for people to legitimately consider. Most companies are not forthright in any way about business conditions... they cook the books till they go belly up and then leave everyone high and dry. It is nice to actually get a heads up from the people working the issues.

Is it always the same note? Are there any other controller functions going on in that track? Graphically it "should" show in most cases, but there are situations where it may not. The event viewer would be the best place to search, especially for a note ON missing the associated note OFF. Quick edit, if that is a downloaded track, the event viewer would also be the place to strip off non-note events that may causes issues.

I am not sure if this is still an issue, but at one time the limiter indicator was reversed (it was lit, but actually off), so that is another thing to check. Synth presets (for any synth) are something to always be cautious with. They are loud, wide, and full spectrum many times so they sound great solo. When putting them into a mix, you almost always need to address all three of these factors to get them to fit properly. AD2 was one VSTi whose presets blatantly went into the red (and some by a lot), so a good safety tip is to have a limiter on the Master buss that doesn't engage till -3dB or so to protect your ears from that shock, especially when composing and inserting new elements.

1903 has been noticeably faster using xcopy and robocopy, but not via Windows Explorer. I have done composition with the Realtek chip for some time, but not all programs support WASAPI yet. For those few that require ASIO, ASIO4ALL has vindicated itself, and buffers can be set lower than they were in the past.

Kontakt 6 also has no number with it, it simply reads "Kontakt" and there are no multi-out variants like Kontakt 5 had.

I habitually type on my phone, so replies can be a bit curt at times. The first paragraph is more an extension of what Imu2002 said, where passing on a signal should be what you want from the start, not fighting with mirror EQ (complimentary boosts/cuts to prevent collisions). HPFs are common to strip the low end off most tracks (they add mud to the mix, as well as energy), but LPFs can serve the same purpose (if everything is bright, they combine to detract from what is supposed to stand out). Choosing instruments to mix which do not collide from the composition stage makes mixing them easier. If you google "EQ Cheat Sheet" these two come up that have nice overviews of components for common instruments: http://blog.sonicbids.com/the-ultimate-eq-cheat-sheet-for-every-common-instrument https://producelikeapro.com/blog/eq-cheat-sheet/ There is no limit on EQs that you can use, or compressors for that matter (slightly compressing vocals twice is often more effective that one big jump), so EQ (for content scoping)->Compressor->EQ (for mixing) is common, but never absolute. As to the second paragraph, reverb (and delays) can cause grief since if inserted too early, then the follow on processing will process reflections the same as if it were the main signal (which they are not). Most "presets," especially on synths, are often loud, cover much of the frequency spectrum, and are swimming in reverb... all three of those need to be addressed before even considering mixing it with something else. Maybe another way to explain this would be an extreme example... reverb is to replicate "environment" since most DAW practice is to strip that up front to facilitate mixing, then put it back in at the end... if a signal with heavy reverb is passed into an amp sim, it will make a mess of things. The reverb is more how the final signal/mix will sound in the "created environment." Again, there are definitely no hard/fast rules to anything, but a lot of the above is to make mixing easier, and the result cleaner. Lastly, if you have never seen this video before it is worth watching. Dan Worrall does an incredible job packing a lot of information into 10 minutes, and explains the reasons why as he goes. It is a video for FabFilter Pro-Q, but what he is doing applies regardless of the FX used.

This advice applies to ALL FX; i.e., be very conscious at all times of what is passing through the signal chain (and showing up in the final mix). If you compress the unwanted portion of a signal, it takes more EQ to remove it, which can also adversely affect more of the desired signal. Bottom line, be careful of what you are passing, that it will fit the mix (if it won't, don't pass it), and what the next FX is going to do. Time-based FX can do the most damage to an unwanted component, because they effectively "smear" those frequencies down the track, so always be judicious with those (and why they almost always come last).

You could replace "bandlab" and "cakewalk" above with "Noel." AFAIK, it has been consistent throughout because of Noel.

The weakest link will always throttle the entire process, since the end result is an assembly. Think of it like the output audio buffer, but faster; a snare track done in .5 seconds is meaningless if not in the context of the output. You are only going to see max CPU on benchmarks, video rendering, and encryption for the most part. Having a car that does 200mph is meaningless in rush hour traffic; you can only go as fast as the person in front of you.