Mix Challenge

Yes, I know among other things, in your language, the tapan and other even more heterogeneous percussion instruments ...

Your recording technique (or sound system) is obviously interesting but, in the end, only brings you problems.

If we simplify to the extreme, it is better to use a single microphone (I love ribbon microphones and have some) and
therefore to carefully research its position in front of the instrument to capture all the nuances. The real work is there.
Once the right sound has been found (with the agreement of the instrumentalist), it's time for a trouble-free recording.
This applies to all other percussion instruments.

Afterwards, in the mix, it's only happiness. No more worries about phase, stereo, whatever you want.

In French we say "who can do the most, can do the least". Believe in my experience of over 45 years of sound recording.

Franz

White Punk OD wrote: ↑

Mon Nov 29, 2021 20:47 CET

I really loved this task, though meanwhile I know what to improve and what I have missed.
The bell and the buzuki flago sounds are a bit weak, and it takes more automation on the reverb (a bit much of a huge celebration hall).
Which is easy to tackle.

@JamesMusic yours is a truly remarkable mix, I say it is high quality, with great separation and balance.
Practically all mixes have some sibilance in the first verse. Almost all have the second verse louder than I would prefer.
You panned hard, so in the headphones the center is a bit sparse, but great space for the vocal.
The kulning sound and the choir work very well.
What I seem to hear is a bit of midrange overload in the finale, because so many voices and flutes are consuming space there.
- I have a question:
How did you treat the drum4-1/2 sounds? They sound very full with low end, but decent enough to not kill the output meter.

Thanks for your comments @White Punk OD, pleased you like my mix!
Appreciate the suggestions, hopefully if I make it into the next round, I'll have an oportunity to take another look at the points you mention.
Your mix has a great live feel, really feels like it's a performance in a medieval hall, excellent job.

I've just had a look back at the mix, on the Drum4-1 track, I've used a subharmonic generator and Pultec to give it more sub, a load of reverb and some saturation to give some higher harmonics and presence. On the Drum4-2 track it's the same reverb, but high and low passed pretty severely.
I think it fits the rest of the drum balance as the other drum tracks I've made to be quite punchy and tight. I was worried it might end up being too boomy otherwise.

JamesMusic AKA Red Shift Mastering
https://redshiftmastering.com/

JamesMusic wrote: ↑

Tue Nov 30, 2021 12:25 CET

Thanks for your comments ...

Thanks back for your remarks,
and for the explanation on the drum processing, very enlightening and competent!
Now I understand why it sounds good.

Problems with piling up sounds and overload in a finale,
I mitigate this with a stem bus structure, that gets compressed by the "register" (as in: "sorted by instrument group of orchestra"), and works like a loop hole that keeps the balance in check, no matter what the participants in the stem are doing. So if needed, I can refine detail balances later, but the mix can't explode anyway. The master compressor eats around 3dB in this case, because for reasons of expression and intensity, the stem compressor is not entirely strict.

Kidokz wrote: ↑

Wed Nov 24, 2021 16:52 CET

I found out about this project less than a month ago and I really liked the concept. But I don't quite understand the LUFS thing. I took the Levels plugin and tracked the level of lufs. But I think my mix turned out too quiet. Although I use compression very carefully.

Apologies for the waiting time to get an answer to this.


Measuring LUFS for individual signals (multi-tracks) is not the route to go. LUFS ILk, or "Loudness Unit Full Scale, Integrated Loudness (k-weighted)" as it is called in long form, is for measuring the whole mix from start to finish and to see if you're within given parameters. If it's too low, like your entry at -28 LUFS ILk, you just need to raise the overall volume of your mix. If it's too high, like a handful of other entries we've had this month, you just reduce the overall output volume of your mix. For example with the help of a Gain Knob on the summing bus, and then checking with a suitable measurement tool (often recommended on this forum, is the free Youlean Loudness Meter 2).

Here are two articles that might help you. One is about gain Staging and what measurement tools work for that purpose. The other one is for understanding loudness measurements and "Loudness Normalization". Both are pinned at the top of the Production Techniques sub forum.

Gain Staging with VU meter (see post #2)
How to: Loudness Normalization (Manually and Automated)


I hope this helps

Gain staging in digital domain is nonsense. As for a mixdown (24 bit PCM), absolutely don't matter if few peaks go above - 1 dBFS, it's ok as long as You don't clip inside DAW. Furthermore, there are some mix engineers that intentionally clip ADC (not any, but the one that are made for clipping, mostly mastering grade) to get a specific sound. This type of converters have built in clip protection, so even in the 24 bit PCM mix that clip to 0 dBFS won't be any problem for a mastering engineer. Lower levels in the mix are also not a problem as long the mix is done ITB, it would be S/N ratio issue with analog summing mixers. Gain staging in analog domain is very important and there is no general rule because every piece of gear is different. It also depends on dBu calibration on DA/AD. Every analog system is unique and only way to do proper gain staging is to learn Your gear, do some FFTA measurements and listen on good enough monitoring system. The thing about analog gain staging is mostly about timbre control. Driving transistor based gear will add odd order harmonics and the result will be colder sound, while driving tube based gear will add even order harmonics and the result will be a warmer sound.

Hope this post will help some to better understand the process of mixing from a technical aspect.

Cheers, GROF!

If we use those popular channel strips with analog emulation, we must feed them with the correct signal level.
If we use only a floating poing purely digital EQ, then it matters only, how things add up at the master fader, that should be in some healthy position, not going all up or (nearly) all down.
A compressor works in a sensible way only, if we are within its adjustable threshold range. And often it will saturate when beyond a certain output level. Same with all non-linear plugins, like tape sims.
If the DAW uses integer arithmetics, then back to analog mode, and learn how yours is behaving, regarding headrooms.

Sometimes, when recorded tracks have chaotic levels, I run them all through a normalization processor, and set that to -18dB (EZ CD does that very very fast). Then, most channel plugins will feel happy, and the summing/mixing will be easily controlled with faders not too far from zero.
The bus structure has to start off with all bus faders at zero (=unity gain).
My 2 cents, YMMV.

White Punk OD wrote: ↑

Thu Dec 02, 2021 12:48 CET

If we use those popular channel strips with analog emulation, we must feed them with the correct signal level.
If we use only a floating poing purely digital EQ, then it matters only, how things add up at the master fader, that should be in some healthy position, not going all up or (nearly) all down.
A compressor works in a sensible way only, if we are within its adjustable threshold range. And often it will saturate when beyond a certain output level. Same with all non-linear plugins, like tape sims.
If the DAW uses integer arithmetics, then back to analog mode, and learn how yours is behaving, regarding headrooms.

Sometimes, when recorded tracks have chaotic levels, I run them all through a normalization processor, and set that to -18dB (EZ CD does that very very fast). Then, most channel plugins will feel happy, and the summing/mixing will be easily controlled with faders not too far from zero.
The bus structure has to start off with all bus faders at zero (=unity gain).
My 2 cents, YMMV.

I was refering to gain staging on DAW's mixer, not some plugin emulations since there are some rules based on a myths. Most simple and safe way is to leave master fader inside DAW at 0 and not move it at all, just set track faders not to clip in 16/24 bit export. For 32/64 bit floating point export even clipping inside DAW is not problem at all, it has headroom for days...

For example, try to export mix in 32 bit FP with 24 dB clipping. Then put it back in DAW, reduce level for 24 dB and see what's gonna happen.

This is a topic for a dedicated thread, and as already seen, can result in heated debates.


To simplify:

There can be endless arguments as to why -18dBFS, -20dBFS or in some cases even -12dBFS were used as "reference level inside a computer". Keep in mind, this was set up by technicians decades back - and then this was adapted for "analog modeled tools" (well, the majority of them with fixed reference levels). And while you're right that completely digital tools, without any reference level for saturation, running at 32bit or even 64bit floating point won't really matter what you send in/through/render out... (thankfully the times of 20bit integer or even 24bit integer plugins are over - yes, they existed!)

The main point here is the learning aspect. The Mix Challenge tries to teach how to mix, or how to improve your mix.
No matter the equipment you use.


Gain Staging is not "nonsense" in the sense that it offers you a way better per-channel fader resolution (after signals have been properly set up), easier integration with external devices/hardware and easier integration with "analog modeled" tools that use a reference level (fixed or varying). Not to mention that there is sometimes still one rogue plugin that works at "integer mathematics" and not floating point. Rare, but they are still around.

The Mix(ing) Challenges are accessible to everyone. We have participants that work completely "ITB" (in the box), others that use a hybrid setup (ITB/software and OTB/hardware), and we even have participants that completely mix on outboard equipment. May it be a digital console, or a large scale analog console. Here, gain staging (overall concept) is absolutely important to not overdrive devices. Unless you want to do that on purpose for "character", that is.


Exporting is just as important, as the game tries to teach you how to stick with given parameters. Especially modern requirements/regulations.

Your entry can be as dynamic as you want it to be, but overall it shall not be more quiet than -24 LUFS ILk min, neither be louder than -16,0 LUFS ILk max, with no maximum peak going above -1,00 dBTP. At -16,0 LUFS ILk max, you can still have a PLR (short for "Peak-to-Loudness Ratio", if you'd use a different measurement like RMS-to-dBFS peak max, that would be called "DR" or "Dynamic Range") of 15. That is pretty darn dynamic and also offers you flexibility at whatever personal reference levels you work at.



I tried to move away from the "just make things loud, for character, people or streaming services will turn down the music anyway" type of topic. That is not the route to go and is the very reason why we still have the Loudness War debates. Hence the game focusing on the the very basics, before "technical rules" were constantly broken (see the last 15 years of audio technology). A mix doesn't have to be squashed to a square wave in order to be impactful.

Please believe me when I say that a lot of thought went into the overall game concept.

Thanks for the help and tips. I apologize for bringing it up here.

No need to apologize... it was a valid question. Hope it helped you find a suitable answer.