Pro mastering engineer reveals embarrassing mistakes 

Glad you like them! I am enjoying making these

I was going to edit visual drawings for those explanations but the hands did it all

@@panorama_mastering Nah brother, strange gestures and unintentional humor are definitely the way forward. It was unexpected and caught me in just the right moment to send me into a fit of hysterical laughter. Aside from all the great content you put out, thank you for making me laugh. Still one of the best sounds ever. Not me, mind. Just laughing in general.

My pleasure! Glad to have you aboard :)

Thanks man! My favourite at the moment has to be Dan Worralls!

The noise is added before truncation. It allows to recover a signal that would have been reduced to complete silence because it causes the noise samples to move up and down following the signal.

Haha good idea!

Spot pn, I did a video discussing the topologies. ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-sQt-vMEJ0gI.htmlsi=S4t4THI-J-E4Z2Q4

​Have you touched on feed back and feed forward compressors in any vid? It's something not a lot of people are aware is even a thing, and can have a huge impact on the way a compressor reacts, and can change the sound of it considerably even between two of the same types of compressor (even the exact same compressor with both feedback and feedforward modes will react and sound diferant in each mode) . Even if just a small section of a vid, it'd be something I'd be interested in. I know how it works, but you always manage to find some nugget of info I've not heard before, so am interested in hearing your explanation of it. I know it might not be the most important thing to worry about in mastering, but it's still good to know about I think.

Yes, that first time constant the gain reduction is at it’s greatest velocity

Yes! Clippers are a great tool, (hard clippers for short transients)

ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-1ormfTMYfv0.htmlsi=F4r1zdd4dXT13xRs

It would depend on the type of compressor in some cases, This is why some compressors are naturally fast types and others naturally slow. As they all acheive the compression effect in diferant ways, what is causing the attack and releace times to be what they are can vary. For example a super fast VCA compressor is all IC chip based and would rely on capasitor discharge time quite heavily, where as an optical compressor is a light shining on a light sensitive variable resistor, and is inherently slow to respond as a result (they also don't have a liniar or logarythmic responce afaik, at least not for their whole curve), and not only vary their releace curves over time, sometimes quite unpredictibley, they also have a memory! As a result they are program dependant too and will react diferantly to diferant things you send through them... O.o With regards to ITB, most standard compressors that arn't emulating gear would *most likely* be exactly as Nick describes, however there's a bunch of emulations that mimik hardware aswell, including all the diferant way of compressing in analouge. However things like most DAWs stock compressor for sure would most likely be strictly logarythmic as Nick described (and some of the emulations probably are aswell). Back in the analoug realm, they would all somewhat be affected by the capasitor discharge time, but only predominantly if the capasitor discharge time is the "weakest link" and the slowest componant to react to and change within the gain reduction circuit (like they are in most modern super fast compressors). They still have an effect on things even if they are not the slowest thing to react, however it's significance in determining the A and R curve is greatly reduced, as the slowest componant to react in the circuit will impart it's discharge or equivilent charicteristics into the releace time instead (again like with optical compressors where the LDR will get "exiceted" by high or long levels of light way after the light has gone off, and any subsiquent short burst of light will then be effected by the memory of the cell of that bright or long exposure to light which will be long after any of the caps within that circuit take to discharge in most cases I would think).

Depends on the type of compressor and circuit.

(currently my favourite yt channel for mastering btw... keep ay-lee-eye-sing your way to the top)

Haha cheers! Honestly, I have no clue how to say it… even when people catch me on calls correcting me… showing me… i canmt hack it

​@@panorama_masteringAye Lee ass (or ess) ing, depending on which state in Australia you grew up in.

Welcome aboard! M808!!!

Depends on the EQ action. Big boosts/cuts with shelves or using pass filters will shift the peak noticably. Minor EQ tweaks and broad bells won’t do so noticeably

(I typed this out and thought it might come across as rude, I didn't mean it like that I swear! 🤓) 12 dB/oct is a second order filter, 6dB per order, so to speak. 12dB/oct at 100hz doesn't mean everything is eliminated altogether, it means it gets lowered 12dB per octave (down 3dB at 100hz). Of course, the K-filter isn't exactly analogous to any (inverted) equal loudness contour, as that would make the filter very complex, computationally (or electronically). K-filter is easy in that regard, and also "close enough", with its high pass filter and top end boost. And equal loudness contours are just squiggles on graph paper anyway, an average of measurements of a random sample of people. There is very little to be gained by exactly tracing an equal loudness contour with a filter, especially as the use of this filter in this use case only amounts to helping a computer to spit out a loudness measurement. And not to be one of those "the song is all that matters, man" types, but the different number doesn't make it a better or worse song.

That explanation wasn't rude at all, haha! It pretty much clears things up for me. So, basically it’s a straightforward and "good enough" way to filter the audio for accurate measurements, right? I had a feeling that might be the case. I think the visuals of the filters were throwing me off, but now that you've explained the function of a second-order filter, it makes more sense. So, with this logic, at 50 Hz, you're around -12 dB lower from the original volume, and at 25 Hz, it's about -24 dB, and so on. Makes a lot more sense now. Thank you!

I will do! Yes second order is 12dB/oct

Their goal was to make a super simple filter, so they only used two "parts". Various more complex loudness measurement systems already existed at the time that were "not invented here". The loudness measurement runs much faster than, say, the one you get in Sound Forge where you have an equal loudness contour. It was also made primarily for TV where they don't care about response at 20 Hz.

How's this instead; I do a limited drop and give them away for free to 5 people? Would you be interested?

@@panorama_masteringI’d be interested that shirt is great

@@panorama_mastering say less fam.

Maybe!

@@panorama_mastering that would make a cool video!

I am curious if the prering will shift the clipping at all? Time to test!

@@panorama_mastering if you make a video ill be watching :) i also almost always do a little clipping first so id be interested to see the results !!

Good observations! Thanks for clarifying!

If you need to do it, do it. It's better to not do it into anything that shifts phase such as an EQ if needing to do heavy equing or hp/lp etc as it will rotate the phase of your clipped signal and shift the flat tops to be on the side of the wave instead, seens as the phase has been rotated by the EQ etc. Using a HP/LP may or may not increase peaks, as far as I understand it it would depend what way the signal is rotated in phase at the peak, as one way would cause more gain, and the other less. I'm not sure if there's any way to work that out easily when EQing, so it's probably best to be mindful and caucious when doing any heavy EQ/filtering and just do it before the clipper, that way even if it adds gain, the clipper can deal with it, and because it's before the clipper, the clipping doesn't get it's phase rotated too and you retain your flat tops.

Or use Linnear Phase eq.

​@@rossbalch Linnear eq introduce pre ring and so on and so forth. I am sure that he was trying to make a point but I don't get it

Correct hpf before clippers

Mine too!

Exclusive Panorama drip!

Always. Even og lossless master, never know what people will go and do with that file, rather have it on than not

@@panorama_mastering Allright thanks for answering!

Here you are!! ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-1ormfTMYfv0.htmlsi=7VDjfTcK7lsYWXVp

Dan Worral

Glad you enjoyed it!

Yes!!

Amazing! Thanks for sharing

No sweat! I will try!

The purpose of a mastering engineer is to put the final polish on a mix, ensuring that it sounds consistent, balanced, and optimized for various playback systems. While mix engineers focus on blending individual tracks within a song, mastering engineers work on the entire album or project, ensuring continuity across tracks, adjusting levels, applying equalization, compression, and other processing to achieve a cohesive sound. Additionally, mastering engineers prepare the final tracks for distribution, making sure they meet industry standards for formats like CD, vinyl, streaming, etc. Mix engineers could technically attempt mastering, but mastering is a specialized skill that requires a fresh perspective and a different set of tools and techniques. Plus, having a dedicated mastering engineer provides an objective ear and ensures that the final product meets professional standards.

The best and simplest answer is an extra set of professional ears with limited access to make changes. Producers/artists limit mixers with the choices made within the production The mix engineer then limits mastering engineer by sending a stereo file (generally) And the mastering engineer adds the final touches. All the while a new set of ears attached to a new brain all working to make the track as good as possible.

It's the final check in the other room, other system by a very trained ears and final few tweaks to make it sound 5-10% better at the end. If the mix is amazing the mastering engineer confirms that it doesn't need anything and he gets paid just for that. He's the quality judge

Good question! Addressing in a reaction vid soon!

​@@j-station Is definitely the best & accurate answer

Nice Video btw! 🙂

Oh and I mean bit depth, not bit rate of course 😅

Most DAWs use 32 or 64 bit floating point. Dither should theoretically be used due 24bit export. I would be surprised if Anyone could here a difference though.

Thanks, yes you‘re right, so theoretically dither should be applied 🙂

You’re welcome!

1+1=window

Zing!

I’d love to. But I am affraid my effort can spent in other endeavours helping the community in other ways more effectively

@@panorama_mastering Then I am going to have to just keep watching but not fully understanding 😂

Banggg

That’s why you clicked on it and watched it. Smart move from Nicholas

What should I have titled this?

@@brucerobertwayne who said I watched the video 🤣

@@panorama_mastering I dunno 🤷🏼‍♂️ We don’t know other people’s workflow so saying that they are doing it wrong is a assumption. Could be something like “ maybe you’re doing it wrong “

Actually hè say’s “my former self was wrong and I’m completely fine with exposing my own mistakes so other people can benefit from it” but maybe you should watch the video that helps.

Music is about capturing an emotion. If it sounds good and feels good then it is good. It's easy to fk things up when you overthink art.