Things I need to understand about drums

Casey Alien

New member
so I've been making beats for about 3 years and I feel like I'm just not getting it, I don't understand how to control the sonics like you all seem to. I don't have any friends that are into making music the way I am I also don't know anyone who can teach me anything so I've just been learning through go studio tutorials for a few years. Take this track from one of my favorite producers on YouTube for instance https://m.youtube.com/watch?v=9gQ2BrGlrQo can anyone give me any insight on how he got his drums to sound that loud and clear? I only use to studio stock plugins, I don't really understand compression, using my ears on the stock compressor for fl studio doesn't work for me the ratio of say 2 to 1 with a 10 db threshold makes sense in my head but then when I listen to it I can't hear any kind of difference and from what I've picked up the fruity limiter is just garbage. Basically what I'm asking is, is it compression? Simple mixing he's doing? The beat sounds so simple but hits so hard. Am I even able to obtain that kind of sound with fl studio stock plugins? Any insight would be awesome like I said I don't really have anyone to come too about this stuff and I do strive to eventually be happy with tracks I make.
 
Hard to say without hearing your stuff in comparison! From the top of my head, many amateurs lack proper monitoring. Decent speakers in a treated room is a crucial, perhaps combined with a good set of headphones. If you can't properly hear what you're doing, it's hard to nail it.

However, I don't think the beats in that Trauma track are anything special. Very basic 808-like samples, not much groove,, mixed to sound clean and modern instead of vintage. The rest of the music is what makes the track. Maybe you're looking at things in a wrong way? Try making a basic beat quickly, then focus on writing a good song around those beats. That one is much harder than doing basic production on 808 samples..

Listening to that song again now... try to listen how minimal it is. Lots of space in there, for the drums and the other sounds to breathe. Try to imitate that, instead of creating a wall of sound. Use envelopes to make those drums quite short and hard hitting. Think about the arrangement as well, don't put too many things in there.
 
In this mix there is a lot of emotion in the chord progression of the beat/loop, the bass and the vocals, the drums on top add some excitement. It is mostly the warmth of the low end that you like, it is made more hot from the upper highs.

You can make it like this when you have few sound sources in the mix and balance out the dynamics across the frequency range. What that does is that it becomes smooth sounding but also modulation rich.
 
Hard to say without hearing your stuff in comparison! From the top of my head, many amateurs lack proper monitoring. Decent speakers in a treated room is a crucial, perhaps combined with a good set of headphones. If you can't properly hear what you're doing, it's hard to nail it.

However, I don't think the beats in that Trauma track are anything special. Very basic 808-like samples, not much groove,, mixed to sound clean and modern instead of vintage. The rest of the music is what makes the track. Maybe you're looking at things in a wrong way? Try making a basic beat quickly, then focus on writing a good song around those beats. That one is much harder than doing basic production on 808 samples..

Listening to that song again now... try to listen how minimal it is. Lots of space in there, for the drums and the other sounds to breathe. Try to imitate that, instead of creating a wall of sound. Use envelopes to make those drums quite short and hard hitting. Think about the arrangement as well, don't put too many things in there.
see that's one thing I've always wanted to be able to do but never had an explanation. How to get my kicks and snares short and snappy mine always seem to have a tail that other producers don't have in there beats
 
In this mix there is a lot of emotion in the chord progression of the beat/loop, the bass and the vocals, the drums on top add some excitement. It is mostly the warmth of the low end that you like, it is made more hot from the upper highs.

You can make it like this when you have few sound sources in the mix and balance out the dynamics across the frequency range. What that does is that it becomes smooth sounding but also modulation rich.
you're definitely right what I love most is the warmth of the samples low end. And the brightness of the kick hi hats and snares. Is that a finishing touch of the beat? Like saturation on the low and high ends? Or is what you're talking about different?
 
you're definitely right what I love most is the warmth of the samples low end. And the brightness of the kick hi hats and snares. Is that a finishing touch of the beat? Like saturation on the low and high ends? Or is what you're talking about different?

The warmth of the low end is actually captured already during recording, in other words the context of a sound source is highly critical to a particular frequency response of a sound source. You can have a room that causes a cold sound and you can have a room that causes a warm sound. You can then capture that with recording gear that tend to add more of it rather than remove that warmth from it, hence you end up with a particular type of sound.

Don't underestimate the importance of a great recording room that is validated in a great control room. It allows you to capture the sound sources the way you want them to sound in the recording and that is the approach that gives you the best modulation-noise ratio. Hence the beauty...
 
Can you explain the concept of "modulation" as you are using it here, DR?

GJ

It's kind of like the "signal" portion of the signal-to-noise ratio, but I am using "modulation" because it describes the sound effect more, signal is a very abstract term that is difficult to relate/translate into some particular audio quality. Modulation is the amount and quality of sound source information that the signal delivers and can be understood as "vibrational richness" or really loosely as "vibe", in electrical terms it's best understood as voltage - the amount and quality of the information increases cumulatively with higher voltage.

When you take an input track volume fader and move it upwards, with each step higher you get cumulatively higher vibrational richness, or in other words you get cumulatively closer to the natural vibe of the sound source. A vibrational rich mix is one where the sound sources are near their original vibrational richness and these mixes are beautiful. Therefore an incredibly important part of music production is to arrange the music with modulation in mind, using sound sources that send beautiful information, in a room that enrich that type of information, using monitors that are able to properly validate that information, captured with high quality gear.

Now, in a mix, modulations might blend well or not so well. When they don't blend well you get noise from the blending, noise can also come from the background as well as from the sound source itself. Therefore the art is to a great degree to get optimal information into the recording, get that high modulation quality blended with the rest to produce minimal noise, clean up various types of noise sources and alter modulation with stuff like saturation, chorus and reverb in order to reduce perceived bad noise from modulation sources that are not blending so well.

Also notice that modulation exists as both actual and perceived, the same with noise. Perceived modulation can decrease when you for instance have too short attack time on compressors and have the compressors hammering the sound, that hammering distorts the perception of the modulation and the combination lowers the emotional response. On the other hand, when modulations from sound sources are blending nicely you get perceived resonance - increased emotional response.
 
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You are using a musical term in an adaptive way as an audio/physics term. I wouldn't do that, because it confuses people. I would say "signal" when you mean signal.

So, the fader has to be at a point where the recorded sound mimics the dynamic response of the source sound exactly for it to be optimized in the mix? This goes against many years of pro mix practice, creates epistemological difficulties regarding arbitrary measurements of spl, and is so improbable that it borders on impossible for most people to achieve. Interesting.

GJ
 
So, the fader has to be at a point where the recorded sound mimics the dynamic response of the source sound exactly for it to be optimized in the mix? This goes against many years of pro mix practice, creates epistemological difficulties regarding arbitrary measurements of spl, and is so improbable that it borders on impossible for most people to achieve. Interesting.

GJ

No. One aspect is the sound source isolated from its mix context, that sound source is captured with a certain amount of original information. Another aspect is many sound sources combining to form the mix and you now will have less amount of original information per sound source because you have a limited amount of headroom within which to fit the sound sources and because the sound sources now fight for attention too and fight/mix up each other's original information, the brain can hence perceive even less of the original information. So that is how it works, actual and perceived original information is of importance and combined with the cumulative nature of voltage you get cumulatively less perceived original information with each sound source you add on top of the other ones. In mixing what you want is to maximize both up to a point where you can balance in other mix quality factors too without ending up with too little original information. Then there are additional aspects to this too, for instance that you want more of the original information from say the lows than the mids and highs on some sound sources and what original information you want to have in the mix will vary per sound source. So in mixing you want max amount of total original information to work with and then the art is to distribute this information optimally in the context of the mix and in the context of other mix qualities you want to have.

In order to then be able to produce good mixes every time no matter the recording, you have to understand the recorded content from the perspective of A) what is the quality of the original information of each sound source and B) in the context of the arrangement how do you approach this in order to end up with enough of the original information and get all of your other mix qualities in there too.

So in practice that means you need to segment the recording and approach each segment based on what's going on in that segment. For instance a verse can be very non-dense and then the chorus comes and the arrangement now turns 500% more dense.

You can easily reduce the density when you have high quality of original information reducing it down to, in pro recording you might have that, you might for instance have really sweet core elements like vocals, bass, kick and snare. It becomes tricky when the arrangement is dense and in the arrangement you have low quality original information too. In this case you should send the mix back to recording, but when that is not an option it is probably a good idea to bring in some samples that provide the kind of original information you need.

When it comes to the sound and perception of a mix, there are certain things that are impossible to achieve. It is for instance impossible to turn a poorly sampled piano into the sound of a real grand piano, it is impossible to get the effects of a sweet pedal steel without a pedal steel, it is impossible to end up with a non-harsh sounding mix when it is full of noise, it is impossible to end up with the warmth of a great sounding room without such a room (warmth comes from sound encapsulation, think both recording room and control room) and so on. Recording, mixing and mastering is to a great degree about ensuring you have the original information that you need, that is way more important than all of the mixing and mastering techniques out there. Most underestimate the importance of this.
 
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So, how does the human ear (set of ears, actually), account for this information density regarding a live performance? How about live vs. recorded? Will the brain process a true stereo (or better yet binaural), bit-perfect 44.1/16 recording of the London Symphony Orchestra from the third row center, differently than it processes listening to the London Symphony Orchestra live from the third row center, all other factors being equal?

GJ
 
So, how does the human ear (set of ears, actually), account for this information density regarding a live performance? How about live vs. recorded? Will the brain process a true stereo (or better yet binaural), bit-perfect 44.1/16 recording of the London Symphony Orchestra from the third row center, differently than it processes listening to the London Symphony Orchestra live from the third row center, all other factors being equal?

GJ

There are two important factors to this. When we assume that the stereo playback is re-producing the audio at the same original information density (in reality it is way lower) as the audio played out straight in the hall, the sound coming from the two speakers are now travelling more focused, this in turn "matches" with the geometric structure and surface of the room much less because the audio is not distributed as naturally and evenly against the surface of the room. This means that instead of the surface having a geometrically beautiful application effect on the audio, it becomes more distorted so you get an output frequency response that is reflecting the original information much less accurately, how inaccurately depends also on your position relative to the surface and the speakers, but this is what makes it feel more cold, it is a bit similar to adding distance to the audio.

In terms of the impact of the room on a recording vs. live, there are two important factors to consider:

A) The smaller the room, the more difficult it becomes to position the sound source and the ears optimally relative to the surface
B) The surface area vs power level determines the "wetness" of the room on the audio

So this means that with a larger room that reflects the audio being played at an optimal power level (with surface properties providing a highly dimensional geometric structure) relative to the surface, you are naturally increasing the recording performance of the sound sources (for each and when combined), you simply get a warmer more natural resulting frequency response of each recorded sound source.

When you record with this in mind, using appropriate gear for the capturing of the audio, you get a more beautiful sounding final result. In combination with the acoustic properties of the control room it becomes a very significant differentiating factor, it simply has much greater depth of emotion because truth is high emotional depth.

If you want an optimal live music experience, you should measure the size of the hall and relative to that draw a circle positioned at the center, with a certain size of it that is relative to the size of the hall. You should then ensure you get a seat that is within that circle, the bigger the hall the bigger the circle. That is the first major factor. The second major factor is understanding the geometric structure of the surface, meaning at what dimensional level the geometric structure of the surface material is at and combine that with at what dimensional level the geometric shape of the hall is. For this you need to account for the floor as being of a variable geometric structure, because it typically depends on whether the people are standing or sitting and how booked the floor seat rows are. Any hall which is like a tent or disc in its shape is very good. For instance Municipal Auditorium in Nashville has incredibly good geometric properties, so if you want to have a great live concert experience, book a seat as closely in the center as possible in that arena on a concert that is fully booked. You will be amazed by how good music can feel.
 
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What if the speaker as a "source" (speaker emulating the source) is in mono, and only one speaker. In the exact same place as the originally recorded sound? And if this mono speaker reproduction is played back and recorded, in stereo, from the third row center. What will be the difference then, all other factors being exactly equal? What about headphones?

GJ
 
What if the speaker as a "source" (speaker emulating the source) is in mono, and only one speaker. In the exact same place as the originally recorded sound? And if this mono speaker reproduction is played back and recorded, in stereo, from the third row center. What will be the difference then, all other factors being exactly equal? What about headphones?

GJ

A mono sound source played out into the room is even more focused/directed, so in this case the sound source is going to match even less with the surface. What I mean by that is that because the sound hits the geometric structure of the surface so focused it means that at a given time it is going to reflect off of the geometric structure at fewer points and hence not utilize the full dimensionality that the geometric surface offers. When the sound leaves the instrument more naturally it is using more of the surface sooner and hence utilizes more of the full potential of the geometric structure of the surface. To understand this you can try to imagine the sound as an infinite amount of tiny laser beams that can either spread out symmetrically from the sound source towards the surface or be directed in some much less symmetrical way. It is all about achieving high degree of symmetry within a highly complex/high dimensional geometric surface/room, so that there is minimal and even distribution of the "crossovers" (when using the laser beam to illustrate it). When you have low symmetry, uneven distribution of- and tons of "crossovers" you get resonances that are accumulating unevenly and hence results in a distorted version of the original information. When the geometric structure of the surface gets more and more complex (the number of dimensions it expresses increases), you get less and finer "crossovers", so visually it looks more and more beautiful also, if you would look at it as a net of tiny laser beams.

When it comes to headphones for audio re-production the same laws apply, meaning for instance that the symmetry of the surface inside of the cups as well as the material of it, is highly important for re-producing as much as possible of the original information. It is also important that they can be fit over the ears in such a way that the position is not "off" relative to the center point of the geometric structure of the cups and the ear. Overall, bigger cups means better sound, when everything else is equal, unless of course you use in-ear buds, which is a different scenario. Some headphone manufacturers like for instance Audeze try to angle the surface so that it is in a 90 degree angle with the ear canal. That works better as long as the symmetry is not ruined by it.
 
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What if our hypothetical speaker was radial, in that it would completely mimic the original source and have completely even distribution of laser beam sound wave source content? As I've said several times "All other factors being equal?"

GJ
 
What if our hypothetical speaker was radial, in that it would completely mimic the original source and have completely even distribution of laser beam sound wave source content? As I've said several times "All other factors being equal?"

GJ

That is optimal, then it becomes more a matter of the room (although the room is always important) as well as the positioning of the speaker and listener relative to the room.
 
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