Lord Tim

My v12 plugins crashed badly too, this is on a clean system with no prior Waves installs too. I uninstalled and put v11 in, and all is rock solid now. There's definitely something funky going on with v12.

This is going to sound snarky and unhelpful (apologies for that!) but... what does it sound like to you? There's really no guide for saying something should be X amount of dB louder or the wave display should look different, it's all about how it sounds in the end that matters, and that'll be subjective to every listener. Now adding to that, this is all down to personal taste for a start, and - probably more importantly - what's right for the song you're doing. As a good example, if you're doing a huge wall of sound tech death metal song or pumping EDM track, your file is going to look like a solid brick and be utterly slammed - and it should be, that's the intent of those styles (generally). Do that to a folk song or a really dynamic jazz piece and you'll get run out of town by everyone because it'll sound terrible. Those styles need dynamics to work. Listening context is pretty important here too. If you're sitting in front of your stereo and appreciating all of the dynamic ebbs and flows, that's great - the extra dynamics are probably going to add interest to the song. On the other hand, if you're sitting in your car in traffic you might find that your quiet sections will go away. This is really a long-winded way of saying "context." What sounds right to you? How will you (or your audience) be listening to it? What is the goal of your mix? If you can answer those questions, then that'll answer your original question.

When you do the @ make sure you immediately type their username right after and then wait for a names list to pop up, then click on the username you're wanting to tag. Just typing it doesn't do anything, as you've discovered. @Teegarden

I found SI Drums extremely loud compared to a lot of my libraries here, so it's most definitely quiet in comparison. I've got a hybrid kit template of SSD and Addictive Drums for my drum template (there's nothing that touches the toms on SSD I reckon, but I don't like the SSD cymbals a real lot, especially for long sections of crash cymbal grooves), and a few custom samples being triggered as well. Everything is run out to their own tracks where I'm running compressors, EQ, etc on it all, and I'm finding it pretty decent levels wise. Try upping the gain a little on each output if you can, that might help a bit. Alternatively, run everything to a drum buss and crank that a little bit (or put a limiter over the top if you're not worried about playing anything in real time due to the latency they typically add to a project) and see if you can match the levels you're expecting.

My vote is to shoot support a message. This obviously isn't widespread or the forums would be on fire right now, so it's got to be something specific to your environment. There's really only so much we can suggest without being a bit more hands on with your system like support can be. These weird outlier cases are horrible to diagnose, but if you can solve it, often there's a good knock-on effect for everyone - if the issue actually does work out to be some shortcoming of the CbB engine and it gets fixed, we all win. But yeah, shoot support a line I reckon. If they solve it, definitely report back here - I'm sure a few of us are keen to know what the solution is!

Great advice right there You might find that you'll get better luck by using impulse responses and an IR loader rather than the built-in cab simulation. But proper gain structure and EQ, as well as a good DI signal in will go a long way to getting a great tone.

Derp. I was right-clicking and selecting Delete rather than pressing the Delete key. Am schooled, and glad for it.

@scook will likely be along to tell me I'm wrong and there is a way (HAHA) but while you *can* select multiple nodes by right-click-dragging around them, if you try to delete them you'll only delete the first one. The info *is* in the page you linked to: ... however, there's a LOT of information on there, so it's very easily overlooked. This is really one of those things that you kind of just pick up over time and incorporate into your workflow, and forget how you even know this in the first place...!

Or, make 4 nodes, and in the section you want lowered, hold down CTRL while you drag down on the automation line and it'll move the entire section.

Yeah, I have to agree with the sentiment here. I think the features that are being prioritised are ones that benefit ALL of us rather than a subset of users. And if I was a dev looking for stuff to do, @Maestro's list is a solid one. For the record, I'm not at all opposed to adding in things like samplers and that kind of stuff - I'd certainly use them, and for those people who rely on that kind of thing, it would be great to have, and would be a fantastic addition to CbB. But this is now free software, and as yet none of the paid add-ons are up for sale. If Bandlab were aiming for a huge market share or to recoup their investment, they're either the WORST businessmen ever (which seems unlikely given what the company is worth) or they have a completely different plan to what paid/subscription software usually does. At the end of the day, why force something to work how you want it to? A Ferrari might be fast, but if you need to take 25 kids to a soccer game, you're probably going to get the job done faster in a mini-bus, even though the other vehicle would run rings around it in a race. CbB is a fantastic DAW that can easily take you from nothing to final master, with some workflows that are superior to other DAWs, and others that are sorely lagging. Just like every DAW. There's no rule you have to use just the one DAW, right? You wouldn't use a multiband compressor to EQ a mix - sure, you can, but an actual EQ plugin is far better suited to the task. A DAW isn't really that different to using a different plugin to achieve the goal when you think about it.

OK, let's try a different approach then and make it relevant: Upload some unmastered music, and someone who does mastering could do an example master and explain their reasoning and method to get there.

It's hard to give any kind of technical contributions without A: hearing the material, and B: knowing the goal. I could easily say "You want to strap a Linear Phase EQ over the master buss and boost 60Hz, and apply 2:1 ratio compression, correct the stereo image, and then add 6dB of limiting and dithering to 16 bit at the end of the chain" but what does that mean? How does that apply to anything? Even knowing half of the picture isn't useful. If someone gave me a really open sounding jazz piece and I mastered it like a death metal release, it would be slammed to hell and sound awful. Understanding the goal is just as important as understanding the steps to get it there.

I can think of 2 great related examples from the hard rock / heavy metal world: Dance of Death by Iron Maiden and St. Anger by Metallica. With Dance of Death, they mixed it (with Kevin Shirley, so not exactly a green engineer by any means) and sent it off to mastering. They got it back and the band basically said they liked the mix from the desk better and put that out as the actual audio release. Now the mix is... fine? But it doesn't have the polish that their other releases have, and on some speakers it's pretty uneven. This is a great example of the band being too close to the product and ignoring the best sonic interests in the end. They've since come around and mastered everything since, mind you. I think everyone even slightly acquainted with Metallica will know what a debacle St. Anger was. In defence of the band. it was exactly what they were going for: raw, ugly, unprocessed... and they got it. No amount of high end gear or mastering was going to save this recording from being anything more than sounding like a very expensive demo. If any unknown band put something like this out, they would be crucified for it and told to go re-record it properly. Regardless of that, both of those albums were massive. On exactly the flip side, you can go massively OVER-produced and overblown too. Spending a year on your snare sound or getting rid of every little drum ring, or correcting timing for every part so it sounds perfect is probably the antithesis of what rock is. Unless you're experienced enough to tell yourself when to shut up and just get on with it, you can dig yourself into a massive sonic hole pretty easily, both with mix AND master. I know I've done this in the past, and I know how easy it was to get myself into that mess. Having an outside perspective is sometimes crucial for a sanity check, and having the ego to be able to accept that check is just as important. We all have the tools to do a pro job ourselves these days, and some of us are lucky enough to have a great sounding studio space to make informed decisions, but it's that last sanity check that's usually the big decider as to who should mix or master your stuff.

Both @Craig Anderton and @marled are on the money - how much you cut is really dependent on the material around it. If it's a very sparse arrangement and you're going for a natural sound, you might leave more of the low end in there (getting rid of any problem frequencies like rumbles or proximity effects with certain mics, etc.) but in a dense mix, you can certainly stand to cut both male and female vocals a lot higher and more aggressively because you won't really hear that cut in context, and it'll clean things up considerably in the mix.

Just adding to what John is saying, while you might have a fundamental frequency of a voice or an instrument audible, sometimes you'll find that once things are in a mix, certain frequencies - even the fundamentals - start to sound wooly because they're sharing frequencies with other instruments, or even more of the same instrument once they're layered. For example, I do a lot of metal and typically a rhythm guitar in metal is a pretty chunky sound. Layer 4 rhythm tracks together and you get a huge build up in the bass frequencies that you need to dip so it all sounds balanced again. Then you add bass guitar and kick drums and suddenly it all sounds wooly again. So you roll off the low end to make way for those other instruments. By itself, those guitars sound super thin and weak, but your ears are going to fill in the missing parts in context when it's combined with the other instruments in the mix. Vocals are the same. You'd be surprised at how much you can roll off of the low end before you start to notice anything going missing once they're in the mix. This doesn't mean that at exactly 300Hz (or whatever) you cut everything off entirely under that, but you can do a gradual roll-off to nothing so you're not putting more stuff into a mix than you'd actually hear in context. I'm sure I'm not saying anything most people that mix don't know already but when you're first starting out, there's a big disconnect between "this thing sounds good" and "this thing sounds good in context." It may not be "faithful" to what's down, but I can guarantee it'll sound a lot better and more professional when you find the space for each instrument and get rid of the stuff that's not necessary in a complete mix. Context is the key. There's some really good advice in this thread that I agree with. My 2 cents: If you're doing vocal prominent music (pop, etc) then start with the vocal, give yourself a decent amount of headroom for that first, and then bring up all of the other instruments around it. If you're finding it's starting to peak out then drop every track down proportionately. Then start carving things out with EQ, and evening out the peaks either with a compressor or clip gain envelopes. You'll likely find the vocals are very dynamic, so certain words or attacks into words will really crank up the levels, so getting into each clip and just turning down the peaking parts will give you a much more even result and make it a lot easier to slot things together, assuming there's no gear related issues making your life harder than it should be.

One additional thing to try is a different offline rendering algorithm too. Some work better than others, especially when there's tempo/meter changes involved. I've been having pretty good luck with Elastique myself, although others have mentioned that tempo changes can really screw up the transients. AS definitely needs some love, but if you know how to deal with its idiosyncrasies, it's pretty powerful.

Yeah, it's time consuming but getting the transients right first is crucial to getting good results and it does mean a bit of manual grunt work moving or deleting markers. That said, though, if you're just playing it back while the markers are active, you'll be hearing the lower quality online stretching algorithms. If you bounce all of those tracks to clips, it locks everything and uses the better quality offline algorithms. You might find your results sound a lot better, even with rogue markers.

Yeah, that's super weird - obviously not typical behaviour. Not even deleting AUD.INI helped? Out of curiosity, on one of the clips that shifted, if you right click it and choose Revert Clip(s) to Original Time Stamp, does anything move at all?

CD was always a risky thing to take to replication if you didn't do it on pro grade gear because the error percentage would often make them reject it, and you'd have to send another copy. And, aside from sending BIN/CUE (which works, but doesn't have any CRC checking in case of a transfer corruption), there was no real way to send a CD online, so you were stuck with posting it or hand delivering it. DDP solves all those problems in one fell swoop. I've only had one single image come back as rejected since I changed from master discs to DDP and that was my own dumb fault for making a typo in the UPC/EAN and not doing an error check before I sent it off. Derp. ? But human error aside, it's been bulletproof.

Yep, agree. I've lurked on the REAPER forums since the very beginning and seen it grow over the years into a truly fantastic DAW. It absolutely does NOT click with me at all, though! HAHA! I spend more time just wondering why things don't work how I feel like they should rather than doing any actual work on it - and in REAPER's defence, that's *my* failing to a degree, there's lots of people who work super fast on it, but I'm definitely not one of them. CbB just fits how I like to work.

Yeah, they absolutely do! This is kind of how the Console Emulator works in the Pro Channel in CbB. By itself it seems like it does nothing, but throw that over 30 or 40 tracks and you really start to notice things sounding different - the subtle distortion, especially on panned tracks really can widen a mix out. It's kind of the idea behind Harrison Mixbus, in fact - that's made to sound like CbB does with Console Emulator engaged on every track by default. Now I'm not saying REAPER has console emulation or anything like that running, but just the subtle gain differences in both the rounding and the pan law can really add up over a couple of dozen tracks, so I'm not surprised to hear that. Of course, it's rudimentary to match that in CbB just by adjusting track volumes, etc. so it's not inherently better as such, just different.

For audio discs, the best way is DDP. But that usually requires mastering software to create or burn from the DDP image. The next best thing is BIN / CUE files. BIN is the audio data, CUE is the track list, and they go as a pair. You can rip a CD to BIN / CUE using http://www.anyburn.com/ To create the files: http://www.anyburn.com/tutorials/copy-disc-to-bin-cue-file.htm Send both of those files to your friend via WeTransfer.com or something similar, have him save them to the desktop and download AnyBurn. Then use this method to create a disc on his end: http://www.anyburn.com/tutorials/burn-image-file.htm (And choose the CUE file as the source image)

Well, do a better test than me so you DO know. Then show us your findings. I'll stick with scientific tests rather than subjective "my ears don't lie lololllol" tests any day.

5 minute test then: Loaded up a 1KHz sine wave in CbB, 1 second on, one second off, duration of 10 seconds. MCompressor was strapped over the FX bin at the default settings except for the ratio that I set to 4:1 so you can hear it better. Exported this as a WAV. Replicated this exactly in REAPER and loaded the WAV I exported in CbB and flipped the phase. Results: The threshold kicks in exactly in the same place, and the difference between the two is consistent with what I saw with the rounding errors I saw above. If you're hearing anything different then it's all placebo. EVERYONE's ears lie. It's entirely subjective. Null tests like this show us up for the fallible humans that we are.

Had a bit of time before a session tonight and I did a quick test. NOTE: CbB Pan Law set to 0dB center, sin/cos taper, constant power. Internally I'm using 24 bit depth, not 32 bit or 64 bit double precision. First we'll do some Self-null tests to ensure CbB isn't creating inconsistent results that don't null with itself, just as a control. ---------------- CbB Self-null test 1: Input files: 1KHz Sine - 24 bit, 44.1KHz stereo White Noise - 24 bit, 44.1KHz stereo Sine Sweep - 24 bit, 44.1KHz stereo Drum Loop - 24 bit, 44.1KHz stereo Output: 24 bit, 44.1KHz stereo files, all exported seprately. Reimported back into CbB along with the original input file and the phase inverted Results: All files entirely nulled to silence. ---------------- CbB Self-null test 2: Input files (all 24 bit, 44.1Khz mono): Kick Ride Snare Bass Hats Volume for each track set to -6db. Panning set to center. Output: 24 bit, 44.1Khz stereo mixdown Reimported this file, inverted the phase of this new track Results: Output meter showed -136dB maximum, or practically inaudible difference. I would expect that to be lower or 100% using 32 or 64 internal bit depth. ---------------- CbB Self-null test 3: Input files (all 24 bit, 44.1Khz mono): Kick (panned center) Ride (panned 100% L) Snare (panned 50% R) Bass (panned 50% L) Hats (panned 100% R) Volume for each track set to -6db. Panning as above. Output: 24 bit, 44.1Khz stereo mixdown Reimported this file, inverted the phase of this new track Results: Output meter showed -136dB maximum, or practically inaudible difference. I would expect that to be lower or 100% using 32 or 64 internal bit depth. ---------------- CbB Self-null test 4: Exactly the same setup as test 3, except MCharmVerb was in the FX bin on the kick and snare tracks with the same settings (see attached pic for settings). Output: 24 bit, 44.1Khz stereo mixdown Reimported this file, inverted the phase of this new track Results: Output meter showed -136dB maximum, or practically inaudible difference. I would expect that to be lower or 100% using 32 or 64 internal bit depth. ---------------- CONCLUSION: CbB completely nulls with itself for all intents and purposes - as expected - even with MCharmVerb and panning applied. ------------------------------------------------ REAPER vs CbB Null test 1: Input files: 1KHz Sine - 24 bit, 44.1KHz stereo White Noise - 24 bit, 44.1KHz stereo Sine Sweep - 24 bit, 44.1KHz stereo Drum Loop - 24 bit, 44.1KHz stereo Output: 24 bit, 44.1KHz stereo files, all exported seprately. Imported the CbB exports from CbB Test 1 for each respective file and inverted the phase. Results: All files nulled to silence. ---------------- REAPER vs CbB Null test 2: Repeated CbB test 2 but inside REAPER, same source files, same settings. Output: 24 bit, 44.1Khz stereo mixdown Imported the CbB export from CbB Test 2 and inverted the phase. Results: Output nulled to -48dB maximum. Not completely silent, but close to silent in a real-world situation. I would suggest this is due to rounding differences with the 24 bit file and the internal bit depth of REAPER mixing engine. ---------------- REAPER vs CbB Null test 3: Repeated CbB test 3 but inside REAPER, same source files, same settings. Output: 24 bit, 44.1Khz stereo mixdown Imported the CbB export from CbB Test 3 and inverted the phase. Results: Tracks panned away from center did NOT null with the default REAPER settings. Overriding the project defaults for each track to dual mono reduced the difference by 6db but the files did not entirely null, and no settings would achieve this. ---------------- REAPER vs CbB Null test 4: Repeated CbB test 4 but inside REAPER, same source files, same settings. Output: 24 bit, 44.1Khz stereo mixdown Imported the CbB export from CbB Test 4 and inverted the phase. Results: Obviously this will not null as per Test 3 because the panned tracks weren't correctly nulling due to the differing pan laws. Reverb decay did not null on the stereo sides, in a similar way to regular tracks panned off center did not null. ---------------- CONCLUSION: When tracks are panned center CbB nulls either completely or very close to 100% with REAPER. The minor difference betwen multitrack mixes can be down to rounding (ie: the exported files were 24 bit / 44.1Khz vs an internal 32 bit or 64 bit mixing engine, so the more tracks imported, the more there will be differences as they're mixed together). In modern pop/rock/metal/hiphop/etc music, apart from very quiet sections or the end of reverb tails, in real-world use, this is going to be practically inaudible. When tracks are panned off-center, the results don't null for any tracks not panned to center. This is due to differing pan laws. This will be audible for those tracks only, even in a modern mix, and can be compensated to match by raising or lowering the track volume. When there are stereo effects added, the results don't null for the amount of sound on the stereo part of the material. For time based effects, this can make the tails seem either louder or quieter depending on the settings and the width of the effect, and will definitely be audible in a modern mix. ------------------------------------------------ Bonus test: Importing the exports from CbB and REAPER into Adobe Audition matched the results seen above. ---------------- My take? I stand by what I said about pan laws being the biggest difference between how these DAWs sound. That doesn't make either one sound better, but just slightly different. The more tracks, the more stuff is panned, the more differences there will be. If I had more time I would repeat this with 64 bit settings to rule out rounding errors with the 24 bit stuff I was using, so don't take this as anything like a definitive scientific test. I'm sure others here could do it better. Ultimately, if you're finding that one DAW is dramatically better than the other then more power to you - use that DAW. If having slightly louder reverbs and some tracks a little differently balanced in REAPER when you're doing an exact apples to apples comparison with CbB, you'd probably get more benefit by actually listening to what you're mixing rather than comparing the differences between each DAW. Can you hear that -50dB track difference? I doubt it. Is a slightly louder reverb going to sound different? Sure. Will that negatively impact your mix? If it does, perhaps practicing mixing more than splitting hairs might be a good option. Reverb settings as used above: