bitflipper

I was about to buy a moderately expensive sample library, not because I needed it but because it's just been awhile since I've had a new toy. But this morning, with mouse hovering over the Buy button, I abruptly changed my mind and decided to have my real piano tuned instead. Although it's not a new toy (I bought it back around 1984), it'll feel new because I stopped playing it 25 years ago after it was last moved. I've had no desire to play it because it sounds awful, having never been tuned after that move. But I remember that it used to sound quite nice and once was a joy to play. Now it's just a nice-looking piece of furniture.

So are you able to insert Breverb into a new project and it works OK? It's just old projects that crash? Maybe it's a case of Breverb (or a dependency) no longer located in the same path that it was previously. The version of Breverb that shipped with SONAR was actually a different version than the one shipped from Overloud. That's because the version licensed with SONAR would only work within SONAR, and Overloud graciously offered SONAR users a free crossgrade to a DAW-agnostic version when SONAR became Cakewalk by BandLab. I just had a look here and I have three different versions of Breverb installed (ProChannel, SONAR and Overloud). Each is in a different location, each has a different datestamp, file size, internal version number and CLSID. As far as the project is concerned, they are three separate plugins, unrelated beyond their names. If your old project referenced a CLSID that no longer exists, that could be your problem. Directory of C:\Program Files\Cakewalk\Shared Utilities\Internal 09/26/2012 11:30 AM 3,247,104 BREVERB 2_64.dll 1 File(s) 3,247,104 bytes Directory of C:\Program Files\Cakewalk\VstPlugins\BREVERB SONAR 11/08/2018 11:48 AM 6,089,728 BREVERB 2 Cakewalk-64.dll 1 File(s) 6,089,728 bytes Directory of C:\Program Files\Cakewalk\VstPlugins\Overloud 11/29/2017 05:08 PM 6,243,520 BREVERB 2 Cakewalk-64.dll 1 File(s) 6,243,520 bytes

Well, they're just going to have to wait until 22/2/22. That's when all the computers in the world reset to 1980 and airplanes start falling from the sky. Trust me, I am a computer expert, and for a small fee I can fix it for you.

Wow, that brings CinePerc down to, um, a meager $450. Yeh, it's an industry standard and all, but that'd heat my home for a month. Personally, I'd be OK with a cool library in the cold, but I live with other people who don't share my values.

Hearing loss is a topic that's come up here many times. Musicians, of course, are more prone to it than the general population. But so are audio engineers, who lose their high frequencies much faster than others (this I heard at a NAMM talk by a hearing specialist). Dave makes an interesting point here, which is that even though he has trouble understanding people talking, he still hears tiny details when mixing. This points out the difference between hearing acuity and critical listening skills. The longer you mix, the more dialed-in you become. I routinely identify flaws in commercial recordings that I've listened to for decades and thought were just fine. Kick beater squeaks, footsteps, traffic noise, people talking in the background - stuff I hear now because I've spent so many hours intently listening for such things. Even as my overall hearing continues to degrade.

Should also note that excessive VLF is quite common, even in sample libraries. Here's the first one I chose at random, from a reputable library developer. The greatest amount of energy is below the usable range (this is not a low-pitched instrument). The presumption is that you, the mixer, will decide how much of it you want to use.

Extreme DC bias in a full mix can mess with your speakers' dynamic range, by continually pushing the driver out or pulling it in and thus reducing its throw. Depends on the speaker, though. Many active monitors have their own DC filtering. As noted previously, just looking odd isn't necessarily a problem. FM waveforms are by nature wildly unnatural and often weird-looking. Also note that VLF content (subsonic frequencies) can look like DC. At what point do very low frequencies become "DC"? I don't know. However, I do know that for most genres, anything below ~45 Hz is mostly wasted energy, and anything below 30 Hz is definitely not needed. Even if you're making bass-heavy dance music intended to be reproduced on large full-range systems, there is a point where it's wasted energy that messes with playback systems and biases your mastering limiter to the point where perceived bass and overall loudness can actually suffer. For this reason, it's a good practice to always have a HPF on your master bus and just get rid of that stuff. If you can't hear it, dump it.

You may want to investigate convolution reverbs, which typically have more nonlinear options than do algorithmic reverbs. The old PerfectSpace reverb that used to be bundled with SONAR could do damn near anything you want with the tail envelope. It's not available anymore unless you have an old installation of SONAR, but there are plenty of similar plugins out there, including the Voxengo version on which PerfectSpace was based. The term "nonlinear" can mean many things. For example, it is often used as a synonym for "gated". If that's what you're after, then any reverb can do it by adding a gate. But if you're using the term to indicate a "bloom" effect, Toraverb from D16 Group does that well. If you're using the word to include "reverse" reverb, I'd recommend tkdelay from tritik. "Nonlinear" can also describe ducking, which the aforementioned tkdelay also does.

Well, that certainly is unusual. Technically not DC offset, though, which results in an otherwise normal-looking waveform whose center line is shifted up or down. This is just half a waveform. Unfortunately, I no longer have Dim Pro installed here so can't investigate further. But my assumption would be that as long as it sounds OK it won't cause any problems beyond looking strange. Note that sounds created using FM synthesis often do look weird.

Yes, with a few exceptions 32-bit plugins work just fine in Cakewalk, and you don't have to do anything special. Just insert them like any other and CW will detect that they are 32 bits and automatically load the bridge. Most of the time it's completely transparent, so if you have a 32-bit plugin you like, chances are you'll have no problem with it. Yes, 32-bit VSTs are "ancient" (sheesh, if a few years makes something ancient, what does that make ME?). However, that doesn't mean they're going to sound any different, because what goes on inside the plugin is still essentially the same.

Are you sure it's actually DC offset? It's possible to have asymmetrical waveforms that look like that, and still be OK. Try placing a HPF on the track and see what that does to the waveform. btw, if it does turn out to be DC offset it's not necessary to fix in an external editor, as Cakewalk has a DC removal feature. It's under the Process menu.

I use this in every mix that has stereo tracks. Granted, technically you don't really need it since Channel Tools does the same stuff and quite a lot more. But if you need to pan stereo tracks the automation will be much easier with this. Especially on anything that's intrinsically stereo such as organ & Leslie, synth pads or pre-panned orchestral sections.

It's a good idea but not a cure-all. SSDs are best-used for storing read-only data such as sample libraries. For recording audio, it's unlikely to make a noticeable difference because when everything's working correctly write speed to the media isn't a major factor. Recording reliability is more about making sure other devices (e.g. network and video cards) and background processes (e.g. updaters, antivirus) aren't interfering with the process. Or... just spend a whole lot of $$$ on a souped-up industrial-strength computer so fast that it can easily keep up even when you are doing things wrong. Not everyone can do that, but fortunately knowledge is free for folks like you who are willing to apply the effort to figure this stuff out.

Zoom way in on the waveform where the pop is. You will see one of two things: either a sudden drop to zero or a sudden inversion of the waveform. If the former, your system didn't have enough time to process the incoming data, in which case increasing the buffer size is the easiest cure. If the latter, you may be overdriving the front end of your audio interface and need to turn down the instrument. There are many reasons for buffer overruns, such as inefficient hardware drivers. If you have wi-fi on this computer, try disabling it. As noted by David Baay above, anti-virus software is another of the usual suspects. Always exempt your audio folders from anti-virus scanning. Sometimes other background processes get in the way, such as automated backups and scheduled checks for software updates. Identifying and mitigating those annoyances is a time-consuming process, but that's what David's referring to when he says "a properly tuned DAW". The amount of RAM you have is a little on the low side, which can sometimes cause problems but probably isn't the issue here. Still, adding another 8GB of RAM would be a cheap upgrade you might want to consider.

Sounds quite nice, especially if you're into 80's synth stuff like Jean Micheal Jarre, Vangelis and Larry Fast. I think this would be useful to anybody playing within that realm (looking at you, Wookie), but especially if you don't already have software emulations of those classic synths (which Wookie does). Listening to the patch demo, though, I didn't hear anything I couldn't (adequately, if not perfectly) reproduce with synths already in my small collection. Admittedly, I do have Omnisphere, the king of pads and sweeps. But you can get a lot of those types of sounds from the free OBXD Oberheim emulator (although I've switched to the not-free but still inexpensive Oberheim 8-Voice from Cherry Audio). I'd suggest looking at Synthmaster next time it's on sale, and picking up a couple of Nori Ubukata's Dawn of Electronic Music patch libraries for it - they nail those classic tones.

Careful not to confuse correlation with causation. Plugins use RAM, too. As they are unloaded, they will release whatever memory they had needed to run. What you are probably seeing is the last plugin unloading after a long delay. That plugin is probably your elusive villain.

What John was suggesting was creating a new project with NO plugins. If that closes properly - as it probably will - start adding in plugins to determine which one is causing the hang. Disabling a plugin does not unload it.

1+1=10 Duh. But to be fair, F + 1 can also equal 10. Depends on how many fingers you're holding up. Two, or all sixteen.

I don't hate lo-fi guys. It is not a moral shortcoming. They are simply misguided souls who need our understanding and patience.

Best product name so far this year. Personally, I don't use loops, having studied the instrument for many years. You could say I was a Rhodes scholar. Mine ended up in a closet for years. I named it Dusty.

iDrums. From IKM, I think.

Sigh. Fortunately, none of the bullet points mention anything about "sounding better", or "sounds more like real Leslie". At least they have their priorities straight.

Technically, the term "pan" only refers to how mono tracks are fed into a stereo bus. With stereo tracks, "panning" is a bit of a misnomer, because a stereo track is really two mono tracks tied together and already hard-panned left and right. The pan slider is actually a balance control, like one you'd see on a hi-fi amplifier. It simply turns down one side or the other. Consequently, the track's stereo image does not change, just one side gets quieter. If you move the slider all the way to the left, the right channel completely disappears, possibly losing important information. This is not a problem for mono tracks, only stereo. A better way to "pan" stereo tracks is to use a plugin such as Channel Tools, which lets you actually position the left and right channels as if they were two mono tracks. That lets you actually alter the stereo image, e.g. making it less wide or shifting the whole thing to one side while preserving both channels' information. Things really get messed up if your interleave is wrong. If a stereo track's interleave is set to mono, the two channels are combined and the track becomes effectively mono. If a mono track's interleave is set to stereo, then the track is duplicated to get (identical) left and right channels at the output. In either situation, the pan control may not behave as you expect it to. A further complication happens when you insert a stereo effect onto a mono track. That turns it into a stereo track internally. Consequently, any panning choices you make should be based on it now being a stereo track. But if the interleave is still set to mono, the output of that track will be treated as mono. Sometimes that's no big deal, sometimes it'll leave you scratching your head wondering why that ping-pong delay is only pinging and not ponging. I know, it's complicated. But you can avoid problems by keeping the appropriate interleave and only change it if using a plugin that you know is going to switch the track to stereo, such as a chorus plugin.