u/TheAuganism

is there a hotkey to toggle the pitch-bend tool in the piano roll? I would love that.

More importantly, the bounce-in-place feature is pretty thoroughly broken and has been getting worse.

Also, different versions of bitwig change how my tracks sound. I have four versions of bitwig installed right now, because certain tracks only sound right in certain versions. It's a nightmare.

All that said, it's still my go-to DAW, but it's quickly approaching the point where I will simply have to give up and go back to Ableton (I'm on OSX). Crossing my fingers that we get a good version soon; I've invested so much energy into bitwig that it would be a shame to give up on it now.

good to know. thanks!

no track grouping

I also miss this from ableton, but you CAN route audio from any track into any other track, so you can effectively accomplish the same thing; it's just a little messier visually.

no looping crossfade in the sampler

agreed this would be nice. I've thus far been able to pick good loop points, but every once in a while, particularly with a little vocal chop, I'll pine for ableton's sampler.

very lackluster stretching algorithm

I find it performs really well for tonal stuff, I think it behaves better for the stuff I like use it for. And there's more flexibility here in 1.1. That said, I miss some of the creative possibilities of ableton's stretching algorithms, especially the grain stuff.

Here's my latest bitwig track, sorta tim hecker meets mr carmack:

https://soundcloud.com/theauganism/chimes-soundcloud-edit

My biggest problem with bitwig is definitely how glitchy it is. Just all kinds of random little things get in your way. In particular, its habit of just not playing bounce-in-place audio in a hybrid track is infuriating.

That said, I STILL enjoy it enough that I use it exclusively these days (for composition, anyway).

I made the same switch many years back, the hardest thing for me to get used to was how patterns are restricted to one instrument, whereas in FL you can group instruments into patterns in any way you want. It just required me to think about my composition differently. Ultimately, this one little restriction gives you a shedload of fringe benefits, so embrace it ;)

ignore the downvotes, this is the truth. went from FL to ableton about 7 years back, and I missed the FL piano roll every time I used ableton. Recently switched to bitwig, which has a piano roll almost as good as FL's and I abuse it all the damn time.

Just something to think about with respect to headroom:

Headroom, strictly speaking in a digital system in the absence of clipping, doesn't matter. It mattered a lot in analog systems, where every step in the processing chain introduced some noise, but we don't have those problems. So when you are told to give yourself headroom or shoot for a certain level, realize that this is mainly to keep yourself from constantly adjusting to avoid clipping. It makes the process of setting your levels less annoying, because you can simply turn something up and you've got enough headroom not to clip.

THAT SAID, you're probably producing electronic music, and you probably want your tracks to sound loud.

I'm going to go on a tangent, and then I have an unconventional bit of advice. If what I've said above is not resonating with you, ignore the rest of this post:

The "give yourself a lot of headroom" approach is equivalent to "don't give yourself a lot of headroom, but when something clips, turn down the master". As in, they yield the exact same results. But people aren't comfortable recommending the latter, and I believe it's psychological: turning down the master makes it really obvious what you're trading off (overall track volume for more volume on an instrument), whereas giving yourself lots of headroom (effectively, turning down the master volume up front) makes it feel like you're adding volume when you turn something up.

So here's my recommendation. It's very unorthodox:

Give yourself less headroom. Turn up your kick to hit at, say, -3dB, if you want a number. Now, turn up the bass so that it's balanced with the kick. Is your track clipping, and you haven't even turned up the rest of the instruments yet? Good. Now, sidechain, EQ, band-delay, do what you have to do to get the track not to clip. Now turn up your next biggest instrument. Your track will probably clip again. Now, do what you must to make that no longer clip. You might be forced to turn down the bass or kick; this is fine. Do it if you have to.

This approach works for 2 reasons:

1) You are getting constant, immediate feedback on clipping. You will get really good at making things fit, and you will realize when you just can't make them fit.

2) Mastering is not magical. In theory, with all the control we have in most DAWs wrt sidechaining, dynamic EQs and so on, a separate mastering step shouldn't be necessary. And during the mixing stage, when you have total control over EQ and level and sidechaining and so on, is actually the best stage to squeeze volume out of your mix. Nobody is crazy enough to take their mixes that far (so far; Four Tet is a notable artist who doesn't get his tracks properly mastered, though his aesthetic is not max-loud), but the principle applies: if you turn your overall volume down to compensate for the fact that your instruments don't actually fit well together, your mastering engineer is gonna hit a wall pretty quickly when he tries to overcome that; polishing a turd and so forth.

OK that's probably more than anybody wanted. Hope this helps somebody.

the recently released version 1.1 has this:

Imgur

which the changelog explicitly says was designed to allow you to disable tracks to save CPU, so I think this is exactly what you're after ;)

Also, doing bounce-in-place to audio will save CPU in the case of synth plugins/instruments, since they consume CPU pretty linearly with the number of notes being played simultaneously, though if you don't bake the effects into your bounce, you still pay for the effects.

I've used ableton live for the past 7 years and am now 100% bitwig.

I think there are a couple features that made me want to like bitwig. I tend to work on a lot of projects simultaneously; to the point where I'll do one project for 15 minutes, then hop to the next for 15, and cycle around like that. As the tracks get closer to completion, that timeslice shrinks even more. Obviously, project tabs are helpful for that.

Per-note pitch bending is absolutely huge for me, it's a really central feature to my sound that I missed sorely in ableton (came from FL studio).

This is almost pithy, but in practice it's probably the thing I appreciate the most about bitwig: that little plus button '+' in the device chain window for adding devices. Things flow so much quicker. This is the thing I miss most when I work in ableton.

But the real reason I use bitwig is that it feels good. Which is the sum total of all the above and a hundred other little things. I just got comfortable early on with it, and every time I open it I feel inspired. I don't even think about opening ableton anymore.

Now, when bitwig gets stable enough for me to completely trust it, I'll use/recommend it for live performance and for mastering too, but it's not there yet. Maybe 2.0?

Add a 'Tool' effect, turn width knob down to 0%.

• Try repitching instead of warping. You'll be forced to get creative to make the timing work, but that can be a good thing, and you won't have the same quality issues as with warping.

• Use FX to compensate for the artifacts created by warping. I'm still learning about this, but one of the ways certain transposition algorithms (e.g. 'Complex' in Ableton) do their thing is by introducing noise to 'fill in the gaps', if you will, which interferes with transients and makes things sound kind of fuzzy and diffuse. Others ('Beat' in Ableton) don't introduce noise and retain transients, but sound jagged when pushed too far. If you're clever with FX, you can compensate a bit here. Try distortion (maybe just on the high frequencies) or very light bit crushing on 'Complex' style algorithms to brighten them up, try subtle reverb/flanger-style effects on 'Beat' style algorithms to smooth them out.

• Melodyne! But always, always try multiple warping algorithms. I've had vocal bits that melodyne could transpose better than my DAW and vice versa. The last song I did with repitched vocals (I'll link if you're curious), I found that bitwig's transposition sounded better than melodyne's, but I still brought the transposed audio into melodyne to adjust the formants and fix the chipmunk effect.

You're not an idiot, buddy!

-"I responded to an FXLayerBro thread" support group PSA

It's funny because there was just another post in this sub

Man I was just responding to your other post like "I have a sneaking suspicion that this is the same guy...", then I read this post and had to laugh.

I like working through these theoretical questions cuz I'm a geek like that, but man two days in a row getting this kind of attitude when I'm trying to help really had me like "wtf internet."

I think it's a workflow thing. This guy for whatever reason wants to have multiple empty parallel racks set up ahead of time. It's counterintuitive to me, as I would just as soon add FX as needed and adjust volume by ear (probably what 99% of us do). As soon as you drop an effect onto the layer, you have to adjust volume anyway, as you say. I guess this gets you in the ballpark immediately?

Personally, in practice I wouldn't do it, but once I realized I didn't know the answer to his question I had to figure it out. Science, etc.

Wow, I'm surprised those work! What gain reduction are you using for the whole rack?

I ask because the formula is supposed to give you the gain reduction for every track equally. So the 3:1 yields -9.5dB, which means if you give each channel -9.5dB it's volume neutral, and the rack itself has no gain reduction at all. But what you have is interesting: the top layer has the most volume (-0db), the next layer has less volume, and they progressively contribute less and less.

Checked for an email back from my buddy, didn't get it yet, but I visited the wikipedia page on the decibel scale and it spells it all out.

the equivalency that we want is this:

amplitude ratio: sqrt(x)

decibel reading: 10 * log10 (x)

That's log-base-10-of-x, since i can't get the damn subscripts to work.

So if you want an amplitude ratio of 2:

2 = sqrt(x), x == 4

plug 4 into the decibel reading equation, and you end up with 6.0206.

Which should jive with what we already know. When I used -6.0205999db instead of a simple -6db when summing signals, there was no longer any "slop" detectable at all in ozone5 during the phase test.

So if you have 3 FX chains, you want an amplitude ratio of 3:1, so:

3 = sqrt(x), x == 9

decibel reading: 9.54245

I just made a 3FX rack using -9.54245, and once again got "perfect" phase cancellation.

I'd like a formula to understand the whole thing.

me too. I'll do a top-level response when I hear back from my friend so people don't have to dig all the way through our crud to get the answer.

I don't understand what the purpose of that was. Mine cancels exactly the same and isn't obtuse, so what am I supposed to understand from that that I didnt originally?

Well shucks I thought it was interesting...

Also yours isn't 100%. If you turn the volume up as loud as you can you'll see there is still something getting through. Both with ur FM-4 and my audio so it might be that the PDC/ADC isn't perfect.

might be, but given that the imperfection can be detected within the DAW itself (try opening something like ozone5, anything with a spectrum analyzer, on the master) I'm tempted to say it's not the converter. I've encountered weird things with trying to do phase cancellation for acapellas in both bitwig and Ableton. For instance, if I start playback at one point in the song, I get phase cancellation at one part of the chorus, but I need to start playback at a slightly different part to get phase cancellation at another part of the chorus.

I did my phase test using the organ, for the record. I noticed that FM4 moves around in a way that different instances are always playing slightly different things.

You've done a big edit since my last response and I think my last response will only be more confusing so, let me respond here to some of the specific points you have:

Yea if you have 2 channels its -6db TOTAL, if you have 4 why would it be -6db per channel? Like why is that even an assumption? It's -12dB TOTAL.

If you have 2 channels, you can either do -6db to the whole thing, or you can take -6db to each channel individually; they both have the same effect. The reason your question is challenging is because doing -6db to each input channel is not the same thing as doing -12db to the whole thing!

what? 1&2 summing is 1+2 and 3&4 summing is 3+4 so why are you splitting them up?

Because that makes the math easy. It's a happy accident (well, probably not an accident...) that we have the easy math that "adding any 2 signals together results in 6db gain". So if you can reframe your problem so that you're only ever adding 2 signals together, you can keep using that -6db rule and build a rack like the one I built above without getting fancy with the math.

In your picture, you have a 2-layer-FX rack, which I'll call 2FX, you have a 3FX, and you have a 4FX. The picture I posted is a way to get a 4FX that is volume neutral. The different between your 4FX and my 4FX is that I am only ever doing -6db reductions, but I'm doing them in stages.

You have to do some complicated math to figure out how to add all 4 of those signals together. By summing all of them at the same time, you have to compute gain reduction so that 1+2+3+4 = 1&2&3&4.

However, I've done it in stages; first I do 1+2 = 1&2 (easy, right? -6db). Then I do 3+4 = 3&4 (-6db again). But then I treat 1&2 as one signal, and 3&4 as one signal. So it's like I'm back to dealing with only 2 signals, 1&2 + 3&4 = 1&2&3&4. We get to the same place, but by applying the gain at different stages, the nested racks keep the math simple.

I hope that didn't make it more complicated ;) I can guarantee you that it works, though, I've just done the phase check myself and get 100% cancellation. If you build the rack yourself and see it work, it'll make more sense.

I need -12dB for 4 chains, -6dB is definitely not enough.

You need to apply it after you've already summed the signals. Look at this:

http://i.imgur.com/ZqbAZyw.png

That is 4 Tools running in parallel. -6db on each chain, and then a further -6 applied at the level of the entire rack (i.e., after each chain has been summed).

So it's not quite what you're after, but it does explain why the -6db rule wasn't working when you're trying to sum all the signals at the same time. I've sent my friend an email, I'll let you know when I know how we can get all 4 of those channels to sum to the same original signal at once.

Hey no offense meant, some people ask some ridiculous questions on here.

The reason I was confused (I read your post many times, actually), and trying to figure out what you're asking, is because the 6db is true. What is confusing about it though is the way it all adds together, and I hadn't really considered that before your post.

If you have 2 channels, it's 6db down, easy enough. But if you have 4, you can't just turn them all to -6db and end up with the same original volume. 1&2 will sum to the original signal, and 3&4 will also sum to the original signal, which means when you do 1&2 + 3&4, you get an overall 6db gain.

What I'm having a hard time figuring out is how to calculate the individual per-channel gain reduction necessary to avoid that output gain for an arbitrary number of signals. You'd think the math would be easy, because it's just adding waves, but we're also trying to express the result on a logarithmic db scale.

One way to do it easily and precisely would be to only work with powers-of-2 numbers of FX layers, and to apply gain reduction in stages. So for the 4 FX layer example, apply -6db gain on the rack itself (because of the 1&2 + 3&4 above). You could then stack two 4FXlayer racks into a parent rack and get an 8FX rack, which itself applies -6db, and so on (though I think you'd have to double the number of layers every time for this to keep working).

I have a buddy I can ask about calculating the reduction so we can simply layer as many FX as we want, so you could have 3 or 5 and not have a crazy nesting thing going on.

• edit: powers of 2, not "even" numbers of FX chain to take the -6db rule to the ends of the earth...

I'm not sure if you're asking about how to set volumes so that any number of empty FX chains won't affect the overall volume, or if there's a way to compute gain adjustment for an arbitrary FX chain to get the same overall volume, so I'll respond to both.

If you take a signal and duplicate it (i.e., run an identical signal in parallel), you'll increase the signal by 6db, so it doesn't surprise me that you've essentially found that number on your own. If you turn every empty chain down by 6db, you'll get the exact signal out (until you add FX of course).

As far as FX racks with actual FX on them, there is no generic formula because every effect is free to modify the volume as much as it wants. Best you can do is adjust each one manually I'm afraid. There are a ton of factors that affect the impact on volume. And even if you get an effect with a frequency dependency volume-neutral, a different signal may not be volume neutral. E.G., a low pass filter might have lots of gain compensation for a sound with a lot of highs (say a snare), but if you run a bass through it, your gain compensation calibrated with the snare will be way overkill for the bass.

Hope this helps. Sorry if I misunderstood your question, please clarify if I did!

what taitgr said, but I'll add that you should try using an unprocessed sound as the attack, and have your processed kick come in right after it.

When you add loads of distortion, saturation, etc. you sacrifice the purity of the transients in the name of the texture and overall sound.

You can hear that the kick (especially in 'Melody') starts off with a pretty pure "click" sound. I'd wager that he's got a separate sound (it can be a wood block, a hit hat, another kick, just about anything with a sharp transient) with minimal processing doing that for him.

Your click needs to get out of the way quickly, and your kick needs to come in as the click goes away. Get in there and tune your ADSR envelopes. The click would stick around anywhere from 10-50ms. I'd put the Junkie Kid click closer to 10-15ms.

Give it a shot, makes a big difference!

So wait a minute... you start a discussion thread asking "Why use ADSR?", but you're really only interested in how ADSR pertains to your specific workflow. You're asking "Why should I use ADSR anymore when I can just draw in the gain?"

Sounds like you shouldn't.

All I'm doing is pointing out that there are cases where traditional ADSR is good to have, to, you know, contribute to the discussion you started... but you don't seem to want to discuss it at all.

So you're saying "I would rather draw the envelopes." Great!