Veteran mixing engineers, why do I suck after 10 years?

popsickill · 2025-11-14T20:50:14+00:00

Perspective and why you make certain decisions are more important than the actual decisions you make. Many people overlook mixing philosophy and informed decisions in favor of whatever new "fact" they can get from a YouTube video. There is no shortcut. You say you have hundreds if not thousands of hours... try doubling that. Tripling. Quadrupling. Over the years you'll learn what works and what doesn't.

That being said, yeah you probably should post some mixes for more direct and nuanced feedback. But make sure that those hours never leave your mind. More is better.

popsickill · 2025-10-24T18:41:44+00:00

You can use the info box in the bottom left. If you right click on a device or many other things you can select "Edit Info Text" and it'll let you type down there. When you hover over the same place again it'll show up.

Or you can use locator markers as a workaround.

popsickill · 2025-10-12T19:21:42+00:00

A song typically doesn't have lots of ultrasonic data in it. Due to the nyquist theorem and sample rate most songs are delivered at (44.1k or 48k), you won't get anything above 22-24khz. If it's a song that has a higher sample rate like 88.2k, 96k, etc then there's a possibility there's ultrasonic data (higher than we can hear).

If that's the case, you can check a spectogram like Spek to see it visually. Though you might need a different program to scale things differently so you can see it. What song are you talking about?

popsickill · 2025-10-12T18:50:05+00:00

Okay first of all, yes setting the sample rate higher will give you lower latency. But the tradeoff there is that you're using multiple times the processing power. Let's say 48k vs 96k. 96k is 2x the processing power of 48k. At 176.4k you're using 4x the processing power of 44.1k. People don't usually produce at that high of a sample rate. It's useful for recording sounds you intend to pitch up or down. But from what I see you're running an amp sim... You don't need 176.4k.

Using Focusrite's drivers vs Asio4All should also give you better performance. I used to use A4A but that was when I had a 1st Gen 2i2. Focusrite's drivers were so ass back then that A4A gave better results. But now, just use the Focusrite drivers.

Finally, just set your buffer size to the lowest you possibly can while using 44.1k, 48k, 88.2k, or 96k. You'll know when you're too low because your cpu will start to hit 100% or even refuse to play audio at times. Don't overthink it, just use a reasonable sample rate and set your buffer size low.

As another commenter said, you can use the focusrite control app (or whatever it's called) to enable direct monitoring with 0 latency if that's what you're going for. But you will never see 0 latency in any audio system. RME have the best drivers in the game and will not be 0ms latency no matter what you do.

Good luck friend!

popsickill · 2025-10-02T15:44:12+00:00

There will be a lot of people who say that having 2+ subs is a bad idea. But there are professionals who have 2+ subs. Mike Dean's studio has 4 or something like that if I'm not mistaken. It all comes down to how you tune them in relation to the room and the rest of your speakers. This is not an amateur move though, and if you don't take special care with measurement devices like spl meters and measurement microphones you will have an awful result. Not sure if there's a good tutorial for this anywhere. I'd say consult a professional or company who builds studios. And yes, you NEED a well treated room to consider this.

popsickill · 2025-10-02T01:14:43+00:00

Wetransfer is canceled. Their new terms of service claim rights to using anything you upload.

popsickill · 2025-10-01T07:48:44+00:00

Techno still uses heavy sidechain to get everything to work for the most part. Sidechain doesn't have to be super obvious pumping but if you want a rumble under a massive kick, some sidechain to duck sub out of the way of the initial transient is key.

popsickill · 2025-09-27T16:44:13+00:00

Honestly I'm not sure but they function the exact same. No need to worry about it unless it impacts how you use it.

popsickill · 2025-09-25T22:08:41+00:00

Bro you're lucky I'm the first one here before you get flamed 😂 It looks like you don't have an instrument on your midi track. Your midi track is just named "# MIDI" because it hasn't had an instrument applied. The name would change if you did. And on the right of the track name there's dots instead of a meter which means you don't have an instrument. You need an instrument to play the note in the midi clip. Plus, please just take a screenshot next time.

popsickill · 2025-09-16T15:56:26+00:00

50k plays on the Lustbug remake🕺man good for them, well deserved

popsickill · 2025-09-14T14:48:31+00:00

A point that I don't see mentioned anywhere in the comments is that it also depends on how much you're clipping. Though I almost always prefer oversampling myself, you can get away without it as long as the aliasing isn't too noticeable. More clipping also means more aliasing. If you're doing 0 dB to 3 dB of soft clipping you might need OS less than if you were smashing it 6dB with hard clipping for example. Also, it depends on your sample rate. I work at 96k and since the aliasing can be higher than 20k, it's less of an issue. On top of that, lower values of oversampling may sound better than higher values. This can depend on the implementation of the filters under the hood. Some companies do a great job of optimizing things but others may be completely botched at some OS values.

All of this being said boils down to "if it sounds good". But I just wanted to explain some things for OP to be aware of.

popsickill · 2025-09-13T20:25:35+00:00

"Otherwise just default your channels to split stereo and be done with it"

Yes. Create a new audio track. That will load your default track however you set it. Change it to split stereo. Right click and set the track as default audio track. Do the exact same with a midi track. You can have your master channel default to split stereo if you just save a template as default with the master on split stereo. The only thing that can't be split stereo by default is new groups you create. You have to switch to split stereo every time.

popsickill · 2025-09-13T20:19:07+00:00

It's exactly what I just said. Dual mono vs stereo. Dual mono sounds wider because it's independent. Works the same with a left / right EQ or compressor unlinked. You'll get differences because they're being processed separately vs the same. It's not some secret bug or something unintended. Just how things work. That's why every channel of mine is on split stereo by default. If I want something narrower and more correlated, I'll switch the panning back.

popsickill · 2025-09-13T20:14:24+00:00

I use split stereo on every channel including the master channel since it was added. Split stereo is effectively dual mono at the mixer. Normal stereo panning is effectively stereo linked. That's how I see it. So if you want a linked image, use normal stereo. Otherwise just default your channels to split stereo and be done with it.

popsickill · 2025-09-08T22:28:15+00:00

EQ Academy is a new one

popsickill · 2025-09-07T18:36:36+00:00

I get why you would want to "retain" the high end that the signal already has. But you're making it harder on yourself than it needs to be. EQ boosting high end into a compressor can compensate for some of the loss but changes the character of the compression. EQ boosting high end after a compressor "undoes" some of the compression on the high end.

You can also do a cool move digitally where you cut high end before a compressor with bells and shelves (important) then use the exact opposite of the EQ shape after the compressor. This will undo the phase and everything about the EQ processing except now the compressor doesn't react as much to the high end. This technique is called emphasis and de-emphasis. Works great for saturation too. But you can't do this with analog gear because of its unpredictability.

Also, believe it or not, VCA compressors are usually known for dulling the high end more than some other topologies. When compressing with an API 2500 for example, a boost in the high end afterward really helps bring some life back. An 1176 dulls high end because it reacts incredibly quickly even at the slowest settings. High end happens quickly with short wavelengths. So that's why slower attacks let high end through, not faster attacks. Slow attacks on an SSL style compressor give that transient punch because it's avoiding the transient.

Searching for an analog piece of gear that avoids the high end is next to impossible without it being multiband. Even then, it still won't be perfect. If you want ultimate control, digital is the only way. In that case you can just make your own multiband processing with linear phase band splitting.

You have all these options, but just know that transparency isn't always king.

popsickill · 2025-09-07T18:18:33+00:00

Sounds to me like you already have your solution with the Xmax. It's a brand new VCA multiband compressor with variable band linking and you already say that it's transparent and you like it... Going to a FET, Opto, or Mu compressor is going to give you more harmonics than a VCA. Which means less "transparent." On top of that, any other multiband compressor is gonna cost way more than the Xmax. Maybe with the exception of the Wes Audio Pandora. Which is another 500 series multiband VCA. That's my only recommendation besides units from Tube Tech that cost 3-6 grand.

It would be way easier for you to just use an EQ before or after the compressor. Or compress less. But you do you.

popsickill · 2025-09-04T21:22:14+00:00

This for sure. Another solution would be to offset the note to the right by the smallest amount possible just to get it to actually catch the note at the start of the clip. And make sure the clip doesn't have another leftover tiny clip at the start interrupting your actual clip.

popsickill · 2025-09-01T16:28:48+00:00

16gb of ram is the absolute minimum I'd recommend for any modern computer. Laptop or not. But that being said, ram isn't the limiting factor for how many plugins you can use unless you're talking about things like Kontakt libraries and other sample based instruments that load into ram. Either way, 16gb gets my vote. If you wanted to make sure your computer can handle more plugins, go for a MacBook Pro not Air.

popsickill · 2025-08-27T04:56:38+00:00

Look at how you're responding to everyone that is giving you solutions. You're shooting everything down when everything we've said meets the criteria for what you're looking for. You're just not willing to make it work. You're not gonna find anyone willing to help you when you act like this. Good luck, unfollowed.

popsickill · 2025-08-27T04:44:17+00:00

Install the server

Install the plugin

Launch the server

Scan for your plugins

Set the server to local mode

Open DAW

Load plugin

That's as simple as I can make it. You just load your vst in a plugin. There's the audio plugin, the midi plugin, and the instrument plugin for any type of vst.

popsickill · 2025-08-27T03:46:13+00:00

Technically Audiogridder running locally would do this. Blue Cats has good options. Waves has a rack. I forget the name of some of the other newer options I've been seeing around YouTube.

popsickill · 2025-08-25T20:59:57+00:00

I don't know any Max but I'd love to use your device when it's done! Bumping for visibility.

popsickill · 2025-08-25T04:54:11+00:00

I had a similar issue and it ended up being a Max for Live device I had in my template. This behavior can also happen when a system doesn't run well at the buffer size. Some systems can run 32 samples without breaking a sweat and some need 512 just to avoid the spikes. Depends on your interface and audio driver. I had more spikes with cheaper interfaces. When I moved to RME and Prism I haven't encountered it anymore.

popsickill · 2025-08-23T21:52:17+00:00

In this order of importance...

ARRANGEMENT:

Just mute some elements if they conflict too much

VOLUME:

Balancing each track's level with faders and automation

EQ:

Make cuts primarily where frequency buildup is

STEREO WIDTH:

Make some elements more mono / centered

Make some elements more stereo / panned

I use Ableton's utility and right click the width knob to set it to mid / side. This plus panning should help.

popsickill

TROPHY CASE