Learn ninja skills with music production, mixing, mastering, sound design, song writing, studio design & acoustics.
Warp Academy empowers artists to reach their full potential, create exceptional music, and live their passion. We’re a global, online collective that includes music producers, audio engineers, label owners, sound designers, festival organizers, booking agents, managers, leading audio brands and more.
We hook you up with all the education, tools, and connections you need to create professional-quality music and launch a successful career. You can learn almost any topic by searching our library of hundreds of free tutorial videos and production tools. Join us. We’re stoked to meet you!
Hey man, thank you very much for taking the time of showing us your processes. It is very informational, and there are not a lot of trustable sources out there. In internet, everybody knows everything and nothing at the same time. The work you did on RX as part of the mastering process is something that I don't see people discuss too much, most people would go straight to the DAW. You are very professional, and I look up to your work and career with utter admiration. Keep it up!
I know this video is old, But I put out a song for an artist and few days later it got a good number of views and listens, However not long after I heard a song that had a sound that mimic mine heavily. Literally copied my sound.
@@warpacademy It’s a scary world we are in at the moment, I wish it was the good days where you didn’t have to face the horrors of such disastrous technologies such as AI. It hurts the musicians that pour their heart and soul into their craft.
Hey hey. They're certainly considered the industry standard for professional mixing headphones. They're the most common headphones I see used by mixers and mastering engineers. I love them.
@@warpacademy Vesper, what do you recommend for me to use with the LCD-X for best results for powering the 18ohm cans. Most people just use an Apollo no? DO you believe audio interfaces like the UA Apollo make much of a difference in sound. And yes, this is for more of a hifi listening config with potential to go producing in the future.
Hey hey. To be perfectly honest, any fairly neutral headphone amp will work great. Don't get lost in the details. Just make sure you're driving them from a pro spec headphone amp, like the ones you'd find in any professional audio interface. I use an Apogee Groove, Duet, and Element 24. I also use a Lewitt Connect 6. They all sound more or less the same. Cheers!
@@warpacademy Hey it's Dan again, thanks for telling me about your audio interfaces, I looked up the Groove, that would do the job and portable for on the go listening as well! Thank you :)
Thanks again Drew! We are very grateful for how much you do for the community. Flow really interests me, being a skateboarder also getting into the full flow state is such an incredible experience. There are times where you are so completely present and focused that there is nothing else but alongside that everything just works so easily and feels so natural. This video has just sparked a thought, you know how some productions just seem to stack and write themselves, you seem to grab all the right instruments and drum samples almost instantly, and before you know it theres a song in front of you? Maybe this can be just how the sessions go with enough mental training..? And not just every once in a while. Thanks man, appreciate the content as ever, you are a next level human, thanks for the insights. Ryan
Hey can you take a look to Voxengo's OVC-128, it is a Hard/Soft clipper with x128 oversampling. Can you maybe, do a comparison with other clippers as you did in this video? Voxengo is usually unknown or underrated to producers, but they have GREAT high quality products.
Hey Dallas. Thanks for the tip. In general I have a lot of respect for Voxengo and their tools. I use many of them and love them. I took a look at OVC-128. I can tell you right off the bat, I would never use it. The interface is really not well thought out and it's missing a huge amount of features that make working with a clipper fun and powerful. Look at plugins like Gold Clip or Orange Clip, even K-Clip. They show the transfer function, they have linkable inverted input and output gain, they have a waveform display showing peak information, they have A/B states so you can compare various settings, preset management etc. Clippers are simple devices, easy to make, so many companies make them. But it's so rare to see a company that actually puts a concerted effort into expanding the feature set, building a thoughtfully designed UI, and making the workflow truly professional caliber. Only a handful consider the workflow that an actual engineer would go through with them, and so they fall flat. Others try to look analog (such as the Acustica ones) and therefore they have all the bad drawbacks of limited analog interfaces. When you use a clipper day in and day out like I do, on hundreds of tracks, you need something that you can flick around fast and get results, plus enjoy looking at and using. The only 3 clippers that fit that bill for me are K-Clip, Orange Clip and Gold Clip. I've ditched everything else. They're all so clunky.
Why the hell, math is not working at position no. 2. Wavelength/4 = 2"(0.61m) Wavelength= 8" (2.44m) Freq= 343/2.44 = 140~ HZ Sonarworks shows 50hz pulled back 9db instead of 140hz. Same is the case with position no.3
Theory and reality don't always mesh. I was quoting some of the information in the same way a theoretical acoustician would, but certain assumptions are made, such as infinitely rigid and massive boundaries that are perfectly reflective. In situ, this is not the case. Theory predicts there will be constructive and destructive interference in the bass range, resulting in a large cancellation notch and a corresponding peak. That is clearly the case in pos 1 and 2. Theory also predicts that the frequencies of the interference will shift, and again, clearly that's demonstrated. Don't get caught up on the math. It breaks down because the acoustics formulas involved are not sophisticated enough to take into account all the variables, and also input data such as transmission, TL, absorption and reflection coefficients are never available for all your materials.
You broke it down in such an excellent way, not boring or complicated.Thank you.Getting the monitoring system right is THE single most important thing you can do for your music. As an example, you bought this nice synth with fat bass, but you won’t be able to enjoy it fully because you have some dips in your room…not the synth’s fault I have the ATC 45a, and until I actually treated my room the best I could I realized that the speakers don’t matter that much, if the room is not ideal.With a nice sounding room, some good budget speakers and with a computer it’s possible to create great music.For those mixing with headphones use a Harman correction curve for your model, I can confidently say you won’t regret it.These are the best tips that I learned, after a lot of trial an error and frustration.One of my favorite producers said once that without an accurate monitoring system all those knobs are useless. Nowadays you can correct the low end with multiple subs, AVAAs and Dirac ART, altough the last two very expensive. Genelec website also has a great article on speaker position. All the best.
Glad you enjoyed this one. Thanks for the kind words and insights! Those ATCs are dreaming. I hope you're enjoying them in your room now that you've dialled it in. Cheers!
Hey Louie. Good question. It really depends on how it's implemented. Usually it's rubbish and I avoid it. Most of them will be using a minimum phase filter that will mangle the phase relationship between your side and mid channel, which is certainly a big NO with your bass. If they use a linear phase side filter, then it can work. I prefer to simply use a linear phase MSEQ like Pro-Q3. Or use a really well thought out tool like Basslane Pro from @toneprojects or Monofilter by @NuGenAudio. Cheers!
Welp, I was definitely among those people who were blindly turning oversampling up. That was my biggest take away from this video: to ignore oversampling.
Hey hey. It's always good to evaluate the features you're using from a foundation of understanding what they do. And , at the same time, don't try to fix problems you can't hear. If you're not hearing aliasing, or you're not hearing it as bad, then why attempt to "correct" it? If you do, however, hear aliasing distortion, and don't want it in your signal, then 4X oversampling can work great.
I agree. You have stated the obvious. Frequency domain effects cannot fix a time domain issue, which is where room reflections affect the signal. However, EQ can do a lot to help tame speaker resonances, and assist in putting less energy into a room mode if you use it subtractively. It's certainly helpful in rooms that are as good as they can get already and need a final layer of adjustment.
@@warpacademy Still if the room is properly built - you don't need that. You can't control reverb time with correction eq so you just turn down mods levels bu if the room is reverberative EQ won't do a thing to it.
You're absolutely right that EQ cannot do a thing about decay time. Only acoustic treatment can do that, which is why I have such extensive content on my channel about that topic. It's pretty rare to see a room that is so well dialled in that you couldn't benefit from a touch of EQ. Even if it's to add a house curve. In my control room, I simply use it for broad spectral tilting. I shelve up the LF if I'm mixing quietly for long sessions, and I'll tilt down the HF (per Bob Katz's approach) sometimes. Because, psychoacoustics and equal loudness contours. Also, any control room with even a minimal to moderate amount of acoustic treatment will never meet the criteria for reverberation. You'll get a controlled decay, which is not reverberation.
A single layer of this module won't do much at 30 Hz. But if you layer them, with air gaps, and use absorption of different densities, they can work wonders. They absolutely can control 30 hz and I have the acoustic results to back that up. My control room is made of very similar modules, layered, and using materials of different densities. I have zero issues at 30 Hz despite the fact that my Neumann KH420s cut off at 26 Hz, and the front back axial room mode is 28 Hz. Here's the design of the room: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-5VrG2K_E7qI.html. No fancy pressure-based treatments, no PRDs, no QRDs, no active trapping. Just the strategic use of porous absorption to make a non-environment room.
@@warpacademy That's what you should mention right away))) layers of them with air gaps. I know what you're talking about. i design room with the same principle now. How good is absorption coefficient?
I should clarify that I never said this is a "bass trap". This is a wide band absorber. A bass trap design would integrate a pressure-based treatment approach, graduated density, thicker absorption, range limiters or all 4. It depends on the room, the room cutoff frequency, and the cutoff frequency of the monitoring system. In terms of the absorption coefficients, they are frequency-dependent and vary based on depth and density. I discuss this more fully in this video about porous absorption: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-SSn8HEsG8ro.html. And this video about the wide band absorption modules I use: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-ECazGzutkV8.html Hope that helps clarify. Cheers!
It is true what you say. I've downloaded songs from RU-vid and Spottify, played in my DAW and analyzed. Genre and style was mostly POP and Rock. And the songs was ranging between -6.7 and -9 integrated LUFS. And most of them went slightly over TP into red. I'm not and expert as you are and still learning. But for me it shows that -14 LuFS is not right as per most advice put out there. For me I just want the proper loudness for genre and style, to be able to compete with other songs loudness and most of all to not sound bad. My first try to make a song loud was not great. It sounded horrible. I guess too much limiting😂 It gets better but still struggle with certain songs to get the loudness up and still sound great. Thanks for your videos. You do know a lot about your craft.
Exactly. Most professional studios that are built around acoustics best practices, that is. Which is exactly what I did in my room. Here's the finished room: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-XfO_btDeXjc.html
@@warpacademy Im right now also in the planning phase of my studio. Is it possible to get it touch with you per email, maybe you can give me some tips for my room and the dimensions. I'm planning to make drywall too and make the sound and dimensions as symmetric as possible. Until now I only finished the framework of one wall ( front wall) but it's still easy to change things. Can you help me to plan my room ? Im also willing to pay a tribute. That would be nice! I'm also planning to make some videos and I would mention your videos and erpertice and help in them. That would be awesome.
great, clean and easy build! 1 question, what happens when more dense wool is used? e.g. 48 or 60 kg/m3? Will these absorb more bass frequencies or will they become reflective insteaad?
Thanks very much! Here's my deep dive video about porous absorption: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-SSn8HEsG8ro.html Watch it for a big discussion around density and depth. The acoustic performance of porous absorption depends on both depth and density. So whether it's reflective or absorbent will depend on how deep the treatment is. If you are considering using 48-60 kg/m3 rockwool, then perhaps only use it as a front layer (1-2") and then lower density behind it. Graduated density will typically perform better than a single density.
Hey hey. I have an updated video that includes full details on the floating flooring. Building a Mixing & Mastering Studio - Part 2, Design Walkthrough ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-5VrG2K_E7qI.html
Love the content but can you turn down the lows a little bit in your mic recordings? Your narration is very boomy, it distracts from what your doing in the daw
Nice! Roar is definitely my most used Live device since 12. Those modulation parameters are astounding along side those routing options (M/S, Multiband..). Super glad Ableton upped their Saturation game. My fav Saturator for mixing use-cases is Wavesfactory Spectre.
To anyone stumbling across this and wondering how to fix micro transient issues I really recommend his science of clipping video, in one go clearly taught me how clipping works, and learning how to hard clip has brought my masters to a new level. Never thought I’d see the day where I could cleanly have a track louder than 5 LUFS without breaking a sweat (though ofc it requires a good mix)
Weird recording chain issue. This was a new configuration for me and took some work to get the bugs out with the interface. I tried some de-noising on it, but it was too watery sounding and heavy handed, so I left it mostly intact in the video. Sorted it out in future ones.
Not sure how you built it but usually it’ll sound different when you miss a key step or a parameter setting. Watch carefully and try again. Model all settings on the devices.
Ah yes. It's a big one to wrap your head around. I've got a pretty good handle on the advanced parameters at this point, but it took me quite a while. I'll note down your idea to do a dedicated Limitless video. Cheers and thanks for being a subscriber.
Plugin Alliance FTW. They have all the Brainworx stuff and so many more quality and affordable plugins. Don't forget TDR, Sonible, FabFilter, Soundtoys too!
Totally hear you on this. Did you see the pinned comment where I addressed that? I misspoke in the video and corrected myself afterwards to say to adjust to perceived loudness and not peak level so you’re not experiencing a psychoacoustic loudness jump that would bias you. Good catch! Because in this example the peak level was significantly reduced while sounding louder.
@@warpacademy that was just a comment on my behalf, i did not read anything below the video prior to writing. you know what you do, and its normal to misspeak smtmes
Thanks for the comment! Not a dumb question at all. RX is intense. I’m not using the de clip for any processing. I’m just using it as a visual reference to draw out where I think my threshold should be. It’s just drawing down to show me what peaks may go over it. And then I manually gain them relative to that imaginary threshold.
Dude... you are so clear and well spoken. Perfect tempo ...and the examples and prepp work is just stellar, i learn so much. Cant thank you enuff. 💚 Please keep em coming. YOU are really lifting my mixing game ‼
Melodyne is amazing. I use Studio 5.4, and it's worth the upgrade just for the multi tracking feature. The ability to overlay multiple vocal tracks for a thick sound, and be able to line them all up to sound like one vocal is insane. Great tutorial. All great features. I like to use it as a plug in like you do also. It seems to be more intuitive. As far as tweaking note volumes, I find using the "make quiet notes louder/ make loud notes quieter" feature to be faster, and then tweak whatever notes/blobs need tweaking. Also, the ability to copy a track, delete all the sibilants on one, then delete all the notes on the other is a great way to tame them by mixing the two together. Looking forward to more Melodyne vids! 🤘🤘
Thanks for the kind words and those amazing workflow tips. Gotta get version 5 now. Those features sound epic. That “make quiet notes louder” feature you mention, where is that? I don’t recall seeing it in V4 but it sounds hella useful.
@@warpacademy I wish I could show you a screen shot..I think the past few versions have what's called the "Note Leveling Macro"..if you look at the top of the Melodyne "page" or "screen" at the top center, just above the timeline..all the macros, and effects are in that strip..the note leveling macro is the furthest to the right of center..the first one to the right is the "correct pitch macro", then to the right is the "quantize time macro", then the "note leveling macro". (They don't say what they are until you hover over them).You should have that feature in 4. 👍
That's terrible to hear mate. Yeah, that's their forced shadow subscription. I bet they haven't even introduced any new features or updated the GUI for that price either. They charge you just to maintain compatibility with new OS versions. Lame.
Hi, old school engineer here. Are you saying that up until the digital age (mid-90's and beyond) when records were made on tape on top of tape on top of tape (24 track 2" to 1/4" mix to 1/2" master, etc) that it was bad (due to intermodulated distortion) because the saturation was done on the mix/master tapes in at least 2 generations? I like setting up my mixes to emulate tape, then analog console non-linearities, then tape again to get that old school saturation effect.
Hey hey. Thanks for the question. I'll admit that tape stuff was way before my time, so I'm just theorizing. The concept is that anything that generates harmonics / distortion / saturation can intermodulate. The big question is, can you hear it, and if you can, does it sound bad to you? I'd run a test by playing 2 pure sine wave tones separately into a tape emulation effect, and then bussing them together and repeating the test. Listen for the intermodulation. A ton of records were made on tape, and recording tape onto tape onto tape, so I'm sure you can get away with it. It's just a matter of if you like the sound. Don't be afraid to try it, just be aware of what may potentially happen. Hope that helps. Cheers!
Woohoo fresh Warp Academy. Thanks so much man! Excellent stuff. I really like the idea of pushing the loudness before the limiter. Appreciate the work as always mate, and thank you for the free template!