bitflipper

Another tip: when you are hand-planting drum hits in the PRV, turn grid snap off. Some of them will be noticeably "off", and they can be nudged - using your ears, not the grid - to correct them. One more: if you must quantize something, make it the kick. You can't advance the snare unless it has something to advance relative to. A steady kick sets up the listener's expectations when anticipating the next beat, so that when the snare or ride cymbal doesn't exactly coincide with the kick it'll have the desired effect. And yes, as Cactus notes, you can "play" drums with nothing more than a drumstick and a microphone. Or even just your hands on the desk.

For your existing drum track, try this. In the PRV, click on the left pane to select every snare hit, and then hit Delete. Now record a new snare track and manually play along on your keyboard controller. Do it a few times and you'll gradually refine your timing to not only fit the song but to enhance it. Don't have a keyboard? Pick up something like this for fifty bucks. What I don't recommend is using your computer's keyboard as demonstrated in the video above, because timing isn't the only factor in humanization. Velocity is just as important. If deleting all your snare hits is too scary, mute them instead. That way, they'll still show up in the PRV as a guide but won't play.

The only thing guaranteed to make your drums sound just as fake as quantization is "humanization". Don't do it. Someday, some clever programmer will invent a smart humanization tool, but it doesn't exist at the moment. Inserting random timing swings does not make the rhythm more human. Yes, human drummers are naturally inconsistent compared to a PLL oscillator. But the variations are not random. A good drummer advances or retards hits in order to add excitement or to relax the groove. When hits don't land on the grid with microsecond resolution, it's for a musical purpose. A random number generator cannot do that. If you want more natural-sounding drums, and learning to play drums or enlisting a real drummer are not options, you can still enter MIDI data by hand. You don't have to be a keyboard player, either. Anybody can tap their fingers. Watch this for some inspiration (note: the presenter is not a keyboard player). Here's something similar but using Cakewalk and SI-Drums:

You're in luck. MMultiAnalyzer for 15 EUR at VSTBuzz. Time to check under the couch cushions for change!

Yeh, you're good. Noel was able to come up with a clever fix, and it works. And no, I don't have a magic formula to squeeze 16 instruments into one instance of Omnisphere. It's 8 tracks times 2 patches each = 16 patches.

Maybe you're thinking of the sidechain kludge that lets you show use multiple inputs of SPAN to display two tracks. It's a clunky kluge, though, and more trouble than it's worth, IMO. There are a number of products out there that do a much better job of this, and while not free they are fairly inexpensive. The best of those would be MMultiAnalyzer from Meldaproduction. It's $72 but periodically goes on sale for half that. Then there's the multi-channel version of SPAN, called SPAN Plus, for $45 and does 95% of what MMultiAnalyzer does. Both are very easy to use. Bluecat Audio's FreqAnalyst Multi is similar but adds a couple unique features for $99, and has been on sale in the past but less often than Melda's.

You're right, that really is worth a look. It would certainly be a time-saver for anyone who regularly mixes other peoples' tracks.

Looks like nobody around here uses that controller. That doesn't mean it won't work, though, and I wouldn't advise fixing the problem by buying another controller. Certainly not another from the same manufacturer, since it'll likely be functionally identical to the one you've already got. I assume you're trying to play a soft synth. By default, the output of a soft synth is muted except during playback. To hear it in real time, you have to explicitly tell Cakewalk to let the synth send audio to your speakers. That's controlled by the Input Echo button, as described here. Have a read through that article and see if any lightbulbs go off. If that doesn't help, or isn't the information you're looking for, come back and we can give you some troubleshooting tips.

I wondered about this, too. Omnisphere only supports eight voices, so more than eight outputs would seem superfluous. It has spurred much speculation among the Spectrasonics user community. The best hypothesis I've heard is that the 9th output allows you to route the aux bus to its own output (e.g. a common reverb bus), something Omnisphere had been unable to do if you've maxed out the multi with a full complement of 8 voices. This would allow you to tweak the reverb amount even after you've frozen the plugin. That makes sense, although I'd have never thought of it myself because I almost always do my mixing within multitimbral instruments - whether Omnisphere, Kontakt, Superior Drummer or the TTS-1 - and then route the multi to a single stereo pair of outputs. Treating a multi as a single instrument simplifies mixing. You'd have to be a seriously heavy-duty Omnisphere user to actually need that 9th output. However, such power users do exist, and they are important to Spectrasonics because they tend to make soundtracks for high-profile television series such as BBC nature documentaries. That's good PR. My theory is that one of those VIPs asked for the feature and Spectrasonics granted their request, unaware that it might cause problems for the thousands of users who aren't famous composers.

Nor mine. But I don't have that particular library. In the upper-right of the UI there are three dots. Click on those to expand a few settings, one of which is maximum polyphony. It should be 256 by default. Also check the preload and stream buffer sizes; not sure but I suspect these may be set automatically based on the amount of RAM you have.

I'm pretty sure all of the LABS instruments are polyphonic. Which one are we talking about?

In that case, you're stuck. Frozen means frozen. Maybe you can identify other tracks that can be safely frozen and thus free up resources for the harmonizer. I deal with this conundrum often. My favorite drum synth is Superior Drummer, which is a huge hog. But I like to be able to tweak the drums right up until I'm done with the project. Consequently, I freeze everything else first and leave that one instrument for last. Sometimes my computer just runs out of steam and I have no choice but to freeze the drums, too. After that, if I need to make a change I have to un-freeze, edit, and re-freeze. Just a fact of life until we all have supercomputers.

You can freeze just the audio portion and leave any effect plugins active. Right-click on the track header and choose Freeze -> Freeze Options" from the context menu. Un-check the "Track FX" option. This assumes it's an effect in the FX bin, as opposed to an effect within a synthesizer or sampler. Those kinds of effects always get frozen.

This happens all the time to me, receiving mono tracks as stereo files. First thing I do is determine which, if any, of the files are truly stereo. You can usually figure that out just by looking at the waveforms. What you don't want to do is convert one to mono that needed to be stereo, e.g. drum overheads, synth pads or orchestral ensembles. But individual kick and snare tracks can almost always be assumed to be mono. You can select all of them and use the Convert to Mono utility. Here's the catch: what Convert to Mono does is combine the left and right channels. Most of the time, it's perfectly safe. But if there are L/R phase differences you'll end up with a track that sounds noticeably worse than it did in stereo. To check for that, simply change the track's interleave to mono first. If you don't hear any change, then the track can safely be converted. So why bother converting them in the first place? Panning is why. The pan slider works differently on mono vs. stereo tracks. Trying to get a nice, wide mix with only stereo tracks is an exercise in frustration.

I'd be very surprised if this was SONAR/Cakewalk version-dependent. This is a system problem and as such likely to affect any DAW. Something is eating CPU cycles, resulting in buffer starvation. To verify that assumption, raise the buffer size to the maximum supported by the interface (on my old MOTU it was 4096) and see if it gets farther before giving up. Next, think about any changes you've made to the system since it was last working OK. New software, updated drivers, replaced hardware components. Run LatencyMon to try and pinpoint a driver that's too greedy. Try disabling the network and see if that makes a difference. Try the Why So Slow utility for grins. It's from Resplendence, the same folks that gave us LatencyMon. And of course one possibility that has to be considered is malware, e.g. something that keeps a network connection open to the mothership, or that scans files in the background. That can even be anti-virus software. A diagnostic tool that lists active network ports can identify spyware, including "friendly" spyware such as tools to continuously monitor for malware.

This is where the oldest guy in the room says: "I remember when a 300 Megabyte drive cost $50,000". Have patience.

I clicked it just to increment Larry's referral count. But my DAW is Pace-free and I intend to keep it that way.

Devil's Advocate: gain staging isn't nearly as important in the floating-point digital world as it was in all-analog days. Back then, it was a big deal. Truth is, digital audio is extremely forgiving about such things. It's entirely possible to complete a project without once thinking about gain staging. At least, I think that's so, because I almost never think about it. Granted, I'm sure I do a lot of things unconsciously, ingrained habits acquired over a half-century. But I have also experienced problems with audio quality precisely because I violated a gainstaging principle. Bad on me for being complacent. I don't recommend complete complacency.

This is how all posts should begin, e.g. "I used to be normal but then learned what a DAW was and never looked back". Your point is valid about how normalizing peaks won't make two tracks sound the same volume-wise. However valid, it's not germane to this thread because the complaints are that the feature doesn't work correctly, not about whether the feature should be used. It's reasonable to expect it to work as advertised, and even more reasonable to be angry if it broke a project.

The TTS-1 is sample-based, and like any sample player it does not have the range of sound-shaping you'd get in a synthesizer. But it is entirely capable of accommodating you in this case. There are several approaches you can take. EQ is the most obvious and easiest solution. Any EQ will do, e.g. ProChannel or the Sonitus EQ. Use a low-pass filter, or in some cases a tilt filter. Modifying your MIDI velocity is the second-easiest. Most, if not all, of TTS-1's instruments respond to velocity. Lowering velocity is a good technique for any sampled instrument, e.g. even fat sounds like big drums sound more epic at low velocities. All of the TTS-1's parameters are automatable. I used to manually drop in some CC events at the front of a TTS-1 track to bake its tone settings right into the track. I don't use the TTS-1 for much beyond click and temp tracks, but I can tell you that it's a surprisingly capable instrument if you're patient and willing to experiment.

I'm surprised that so many people normalize. Honestly, I thought it was a long-discredited practice that only existed in SONAR/Cakewalk as a vestigial feature. As an admitted non-normalizer, I'm curious as to what normalizing multiple tracks does. Does it just normalize each track individually, one at at time, or attempt to normalize them as a group? And what kind of corruption occurs? Is it a corrupt file header, or is the data trashed? I'm trying to imagine how multiplying each sample by a constant could corrupt a file, given that mathematically it's no different than using a fader. I agree that it should be nondestructive, though. I wasn't aware that that was the case.

Tracy Collins is the real salesman here. Peter says "here's a new product, cool, huh?". Tracy says "here's what you can aspire to if you practice". He's a true master of the faux-guitar.

Nah, that's always been the case. Nothing new there. What's changed is the dearth of humor and the ability to see it. It's up to folks like you (and me) to help keep humor alive. Like Steven Wright said "I think they should attach pies to the front of trains. That way, if a train runs into something it's at least a little bit funny". Someday, we may look back fondly at the times when guitars were hard to come by and food wasn't.

Insight for $39. I can't recall ever seeing it that low before. Granted, not everybody needs it. There are plenty of cheap or free metering plugins out there. Technically, I don't need it either, and certainly would not have coughed up $200 for it. But it came with Ozone Advanced, so that's how I ended up with it. I'm glad of that happy accident - it's great and it goes into every project here. Caveat: I am using Insight 1, not Insight 2; version 2 doesn't seem to have anything new that I'm interested in.

That's encouraging for ST4 users. Dave was responsible for much of the content in ST2. Question for Peter: will the Sampletank version be capable of all the key-switchable and velocity-triggered articulations of the Kontakt version? How about chording, harmonies and alternate body IRs?