ORIGINAL: pkev

Hi there,

Reading this thread just brought to mind a technique that Classical Guitarists use. What they sometimes do when playing a passage that uses open strings is to damp the notes according to the exact value of the note on the sheet music...I have used the same technique sometimes when recording Bass...

Oh, yeah. Absolutely. If you listen carefully to recordings by top session bass players one of the two things that really sets them apart is very deliberate control over note duration (the other is dynamics control). Any old Motown recordings are classic examples of this. A bass player with a sensitivity to how his/her notes and especially how the silence between them interacts with the drums and other instruments can make the whole song come together as a punchy, solid unit whereas a bass player who plays with loosey-goosey "expressive" timing and dynamics (as a guitar player typically would, for instance) can make the whole band sound muddy and lurchy and slightly off-time, even if all the notes are technically "one the beat."

It's very easy to overlook what a huge role the low end plays in the "feel" of a mix, and very to misunderstand it. One of the hallmarks of a beginning bass player is a tendency to focus on the upper midrange of the bass sound, and to think like a third guitar player. This approach is not necessarily a *bad* thing in and of itself, but the fact remains that the low end is to a great degree bypassing the audience's ears and communicating directly with the hips and the feet and the hairs on the back of their necks. If nobody is paying attention to what's going on in the low end then the audience might *hear* a good performance but it's not going to move them the same way.

Cheers.

Bump, for those of you that haven't read this great thread

ORIGINAL: Dave King

Usually I will use a low shelf before the reverb at -3 to -6dB if the reverb muddies up. After the reverb, maybe some high-end sculpting to get rid of any splashies or metallic sound.

Yep, can you elaborate on this:

Are you applying the low shelf cut to the vocal signal only, before any reverb is applied and then using additional EQ after the reverb? Is this done by using track effects? In other words in the vocal track effects bin you would have an EQ, then a reverb and then another EQ?

Thanks.

Maybe any of the above, whatever works. Often, the reverb is going to be on a separate bus, and may have other instruments running through it as well, so that can affect the decision-making, too.

Sometimes I really like the sound of the vocal and the tonal balance but there will be some ugliness in the reverb. If I eq the vocal then I've changed the frequency balance (which I like). If I eq the reverb, then maybe I can get the best of both worlds. Sometimes I want to keep the "dry" portion of the vocal track intact, but maybe reduce the highs in the reverb send, if say the esses are triggering "splashies" in the reverb. This is a symptom that is usually easier to prevent (before the verb) than to treat (after the verb). But a lot obviously depends on the routing and what else is going on. In any case it's totally possible to use more than one stage of eq, especially after another effect processor.

I guess my whole point throughout this thread has been to kind of point out that step-by-step "recipies" can only get you so far. At some point, you have to use critical and focused listening to identify what's working for you and what's not, and then use those same listening skills to evaluate the solutions. By far the most important step is simply to identify the fact that there is a metallic ringing in the reverb (or too much boominess in the kick drum, or whatever). Maybe it's from a bad reverb, maybe it's eq, maybe it's the reverb revealing some kind of harmonic distortion from the compressor, maybe it's room sound picked up in the recording, whatever.

The main thing is to identify how you want it to sound, and how that differs from how it sounds now. If you can do that, then the trial-and-error required to fix it is easier than typing the question. All of my talk of finding different "nodes" with the eq and "parking" them was really about this idea of learning to hear and think of the sound in terms of raw components, and how they inter-relate with the other sounds and components.

Cheers.

ORIGINAL: Dave King

Yep, would you say that that your approach to identifying and tweaking frequencies of drums is as necessary with sampled drums (as in EZdrummer) as it is with real drums?

Thanks.

Well I wouldn't even say it strictly "necessary" at all. To be honest, the method I suggested in my first post on this thread is not really even something I use, it was just a suggestion as a way for beginners to think about it. I mean, I do often sweep the frequencies and "park" nodes and think about which ones are interesting in good, bad, or mixed ways, but the notions of "thump," "note," and "click" are ultimately kind of vague and rudimentary...

I learned this stuff by trial-and-error, and developed a feel for it long before I developed any theories of it, which is how I think most audio engineeers traditionally learned. Over time, I have learned to recognize that certain frequencies that are for instance important to the kick drum often happen to coincide with certain subjective but consistent perceptions such as "thump," "click," and so on.

Personally, at this stage, I tend to just instinctively hear which frequencies I want more or less of. Over time, being asked how I knew to turn up or turn down this or that frequency has forced me to think about this stuff in ways that have actually helped me out a great deal.

My hope is that anyone who might find it helpful to start to think about different frequency ranges in a methodical way as I tried to describe above will eventually start to identify things between the "thump" and the "click" and will start to create their own approaches, and that the rudimentary sweeping for the "click" or whatever will become unnecessary. Kind of like how a musician who starts out by learning simple chord progressions and scales begins to internalize them and starts to sort of "hear the theory" intuitively, and can begin to play along by ear even to chord progressions they don't know and to songs they've never heard before.

Ask a dozen good mix engineers to mix a song and you'll get a dozen different mixes, same way that a dozen different piano players will all sound different even when playing the same song. Learning little "licks n tricks" can help a player get started, and can even you by for quite some time. But ultimately the really good stuff comes from an internalized understanding that becomes a fluid expression of a unique imaginative vision.

Engineering audio is not as important as creating music, but the same kinds of principles apply, and really good engineers learn to understand and work with frequencies, dynamic envelopes, harmonic distortion, and such stuff the same way that musicians use notes and chords. You're using a technical mastery to create a better subjective experience.

Nobody really cares how well a musician knows the modes, what they care about is the material and the performance, and the psychic/emotional/spiritual place that the music takes them to. A musician with a greater mastery of the modes and scales can often create a richer and more nuanced musical journey or whatever, but the point is ultimately the experience, not the machine that creates it.

Similarly, nobody cares which frequencies the engineer manipulated or what the compression ratio is, what they care about is an immersive experience of the music, preferably one that is devoid of sonic distractions or artifacts that seperate the listener from the music, and ideally one that unobtrusively flatters and showcases the best aspects of the music.

To answer your question specifically, there's no reason to treat sampled drums any differently than plain old recorded drums. Of course, the obvious advantage to using samples is that you can easily flip through them and you may be able to select sounds that are closer to what you want from the start, especially if recording options are limited by budget or other constraints. So it may be the case that samples require less processing, but it all depends on the samples.

Cheers.

ORIGINAL: yep

Compression:

I'm going to do this a little backwards, so bear with me. I'm first going to describe the theory of compression, the whys and wherefores, and then describe how to actually use the compressor. So if it seems like I'm skipping over stuff, be patient. This is a big topic.

Bass, kick and snare almost always call for some degree of compression in conventional popular music production. These are the most critical elements of the mix, and they are also some of the most prone to wide dynamic fluctuations. Modern popular music tends to have a very fat, full, saturated sound, with electric guitars, organs, synthesizers, multi-layered vocals, and other elements that create a "wall of sound." Drums on the other hand have a very "spiky" dynamic sound that can sound wimpy behind these fat, flat, modern electronic instruments, unless you either turn all the other instruments way down or find some way to fatten and even out the drums. Similarly, bass can be difficult to control in a dense mix. The bass is notoriously prone to wide swings in level from one note to the next, in part simply because of the nature of the instrument, and in part for pshychoacoustical reasons that are a topic for another thread. This can lead to the infamous disappearing/reappearing bass line or the equally notorious bass note that would be king. Compression offers a solution to all of these problems, and also offers some opportunities for creative sound sculpting that can really make your music move and groove the way envision it.

There are two basic strategies for using compression that we'll take one at a time. the first and most basic use of compression is to transparently even out the dynamics so that the instrument sounds the same as it did pre-compression, but it maintains a more consistent overall level. "more consistent overall level" could mean a few different things. It could mean that each individual note or drum hit maintains its original dynamic profile, but that the level is more consistent from one note to the next, or it could mean that the individual notes themselves are modified for instance to reduce the peak level so you can bring up the "body" or sustain of the sound relative to the other instruments in the mix. Both of these goals can be achieved with great transparency using modern digital compressor plugins including the Sonitus compressor.

by "transparently," I mean that compression is applied in a way that does not perceptibly change the "sound" of the instrument. This means that using this kind of compression can be somewhat tricky for the novice, because unlike say reverb or distortion the compressor can be hard to hear if done properly, even with quite heavy compression. It doesn't "sound" like an effect. In fact, that's the point-- to make the sound of the compressor transparent. If you can hear the compressor as an effect, you're doing it wrong.

With practice, you learn to sort of "feel" the way the compressor affects the breathing and pulsing of the music. But when you're first starting out, it is critical to use your meters. You need to look at both the track's output meter (the cakewalk "track" meter, preferably set to RMS+peak), and the compressor's gain reduction meter. Make sure you are looking at the gain reduction meter on the compressor-- many have a meter that switches between showing different things. You will use these meters in conjunction to see what you're doing to the sound. I'll describe later how this actually works, but for now I just want to cover the theory.

The second type of compression is what I'm going to call "saturation"-style compression. This can sometimes be achieved without even using a compressor. When you overdrive certain high-quality analog audio equipment, there is a gradual compression and "soft clipping" that occurs before the ugly, distorted clipping sets in. This is especially true of tube equipment and analog tape. The effect is that peaks are limited, and a warm harmonic "fire" or "fullness" is added to the sound, along with a hyped-up sense of detail. An extreme example is of this phemonenon is guitar distortion, and you may actually be surprised at how effective a distortion pedal can be when used on drums in a dense mix. You can often apply surprisingly heavy amounts of distortion to drums before they start to sound noticably affected.

This type of saturation can be achieved with a variety of special "vintage" or "warming" style compressor plugins, and also with any number of dedicated effects or hardware components, including some effects that aren't really intended as compressors at all. This is an area to get creative and play around. Saturation effects can be a subtle, musical way to add new colors and textures to your recordings. Blockfish by digitalfishphones is a good free plugin that can get you started.

Note that whatever type of compression you use, you should make your compressor settings while listening to the whole mix, not to a soloed track. Compression that sounds extremely transparent and appropriate while the whole mix is playing can sound weird and unnatural when you solo the instrument. Make changes in small increments and use the bypass feature of your compressor as a gut-check. Be wary of the siren song of ever more compression. Overcompression can make everything sound slamming, hyped, and loud, but only for short bursts. After a few seconds, overcompressed music becomes fatiguing to listen to and your audience will turn down the volume, defeating the purpose of the compression in the first place!

If you're just starting out, I would recommend tweaking the compressor so that the gain reduction meter "bounces" only two or three dB on most notes, tracking the tempo of the music fairly closely. Then turn up the makeup gain by about the same (2 or 3 dB). Note that this may sound like nothing has happened--it takes practice to hear mild compression. Then stop playback, rest your ears for a few seconds, and BYPASS the compressor before you hit play. Now listen to the uncompressed track and then un-byapss the compressor and see if it's an improvement. The track should feel a little punchier and a little more "present." If it needs more, try and get the gain reduction to bounce 5 or 6 dB the same way, and evaluate your work the same way. Take baby steps and go carefully and it will pay off quickly.

Now the nitty-gritty. Below is something I posted on an earlier thread about how to actually use a compressor and what the different controls are for. Hopefully it helps:


You really need to read up on this and practice a little until you understand how compression works. Every setting on a compressor affects every other setting, and they are inter-dependent. depending on how your other settings are, adjusting, say, the attack setting up or down can have opposite effects.

There are tons of explanations all over the web and in books of how compressors work, but none of them are very clear until you already have a fairly good handle on certain things.

First piece of advice: avoid using your hardware compressor before recording unless you really feel 110% confident in what you're doing. If you record at 24 bit, there is no advantage to compressing before recording, and software compressors will tend to be better than your hardware compressor. Moreover, if you get it wrong at the tracking stage, there is no way to fix it or undo it.

Second piece of advice: Understand the following before you start using a compressor. Depending on how you use it, a compressor can actually INCREASE the dynamic swings in your program material. Use of a compressor DOES NOT guarantee flatter overall levels. This is very important to understand.

Here's how a conventioanal four-knob compressor works (there are also two-knob compressors, but that's another story). You have four basic controls: attack, release, threshold, and ratio (there is also usually a "makeup gain" knob, but that's seperate from the action of the compressor, it's just a gain control that comes after the compression circuit. We'll deal with it later).

Before we start to talk about what those four knobs do, it is necessary to understand how an audio waveform develops. When you pluck a guitar string, or hit a piano key, or strike a drum head, several things happen to produce a complex waveform. I'm going to use an acoustic guitar as an example, but the following applies to almost any instrument, in principle.

If you look at the waveform of a plucked guitar note, there are typically three (maybe four) distinct parts of that individual note that look like different sections in a visual representation (in sonar's audio clips, for instance).

First there is the inital transient (the "attack" of pick on string). This typically looks like a sharp, almost instantaneous "spike" in level, and is typically the loudest part of the note.

Second, there is the main "body" of the note, the soundboard of the guitar resonating with the vibration of the string. This tends to have a fairly even level, and looks on the computer screen kind of like a brick of sound energy that comes after the initial transient spike. The duration tends to depend mostly on the guitar construction and playing technique. This is the "steady state" portion of the sound, the primary "sustain" of the note.

Third, there is the "decay" or "tail" of the note. This typically looks like a rapidly narrowing triangle attached to the end of the "body" of the sound. It is the section where the string starts to lose energy and the soundboard is just vibrating from inertia. it overlaps slightly into the time where the guitar has stopped vibrating altogether, and the final tail of the sound is simply the soundwaves still bouncing around in the room.

The phantom "fourth" section is "silence." Unless you record in outer space, there is no such thing as actual silence, but the quality of your "silence" definitely matters for purposes of compression. We define silence as the point where random noise and room mode resonance takes over and is as loud or louder than the sound of the actual note.

Got all that? good. Here's how it matters to your compressor. Your compressor has a little gremlin in it that turns down a volume knob. That's all. Really. A compressor is just a little guy that changes a gain control really fast. Here are how you give him his instructions:

Threshold-- this is like an alarm clock that wakes up your little gremlin. If you set your threshold at -6, and your incoming audio signal never goes above -6, then your little gremlin never does anything and sleeps through the whole thing. Once you threshold goes above -6, the gremlin wakes up and springs into action.

Attack-- If the threshold is the alarm clock, "attack" is the gremlin's "snooze" button. The "attack" is a delay between the time the signal goes above the threshold, and when the gremlin actually starts doing his job. For instance, if your threshold is set at -6, and your attack is set at 20ms, then what happens is this-- when your audio signal goes above -6dB, the gremlin groans, rubs his eyes and hits a snooze button that lasts for 20ms. After 20ms, the gremlin actually wakes up and starts doing his job. This setting allows 20ms of "attack" (aka "transient," aka "peak") to sneak through before the signal gets compressed. So if you wanted to flatten the whole signal, you'd set the attack to 0ms. If you want to keep the "impact" or the pluck of the guitar string intact before you start squishing the sound, then you simply tweak the "attack" setting to taste.

Ratio-- this determines how much the gremlin turns down the volume. It works in relation to the "threshold" setting. So if you set the threshold to -6, and the ratio to 2:1, then the gremlin will turn down the volume by 1db for every 2db above -6 the input signal went. So if the input signal was -3dB, the gremlin will turn down the signal by 1.5dB, for an output level of -4.5. (Ratios above 10:1 or so can basically be consiered a hard limiter)

Release-- this is the gremlin's quitting time. If you set the release time to 0ms, the gremlin quits and goes back to sleep as soon as the incoming signal drops below -6db (or whatever the threshold is set to). If te release is set to, say, 50ms, then the gremlin keeps working for 50ms AFTER the input signal drops below -6, which can lead to smoother-sounding tails.

So here are some examples:

If the threshold is set ABOVE the level of the BODY of the sound, but BELOW the level of the TRANSIENT attack of the sound, and the attack/realease are set very fast, then the compressor will basically work as a peak limiter, only compressing the initial transient attack.

If the threshold is set below the level of the BODY of the sound, and the attack and release are set somewhat slower, then the slow attack time will allow the transient sound to pass right through and will then compress the body and "tail/decay" of the note. This will create MORE initial impact when you apply makeup gain, and the slow release time will create a longer, more gradual sustain. This is the most "dramatic" way to set your compressor.

If you set a fast attack and a slow release with a low threshold, you can actually create a sort of "inverse" waveform, where the attack is compressed and the "body" of the sound sort of "swells" after the initial sharp compression releases.

If you have very aggressive compression settings with a too-fast release time, the compressor will "let go" too soon, and the decay and silence will pop through as amplified background noise between notes, causing "pumping" or "sucking" artifacts. On certain kinds of dance/club/electronic music, this can be a cool "throbbing" effect on bass or kick or synth pads. On most kinds of acoustic music, this sounds awful.

If you set the release time slow enough, the compressor will overlap onto the next note, and reduce your transients even if you set the attack time to let the transients through, again creating a "pumping" effect, but with a slightly time-dragging feel.

Depending on how you set the compressor, you can turn a sequence of steady quarter notes into short, choppy, staccatto hits, or into a smooth, pulsating pad, or make the notes sound like they're pushing or dragging the beat. You can make a track sound punchy and slamming, or mellow and smooth.

Listen to how the compressor affects the way the music breathes and pulses. Listen for the changes in performance "feel." Don't try to use the compressor as an automatic volume control-- use your faders for that. Don't use a compressor to limit stray digital overs on the incoming signal-- use a limiter for that, or better yet, just turn down your record levels and deal with your levels in the software.

Finally, and THIS IS THE MOST IMPORTANT THING TO KNOW, never rely on "presets" or "recipes" for compressor settings, unless you're simply using the compressor as a hard limiter. Correct compression settings depend of the tempo and dynamic profile of the incoming signal, and on the desired dynamic profile. THERE IS NO WAY any preset-writer could have known this about your material. Presets may be useful for reverb, saturation effects, even eq sometimes, but almost never for compression. Anybody who tells you that the right compression settings for, say, electric bass are such-and-such is WRONG, unless they always record the same player and the same instrument at the same tempo every time.


Cheers.

Part of what you are saying more applies to treating recordings of live audio performances, where you do compress individual parts transparently first, so they have a more consistent level. Its the difference between corrective and creative compression. When sequencing, or using corrected audio, you already know or can set the levels of notes in a part, and its more about changing the dynamics of the sound itself.

You're slightly confusing overall compression/limiting/maximization with compressing individual elements. ? :)

Compression on individual elements is an effect first, a volume maximizer/leveler second.

When compressing a kick alone, its not about transparency, you usually want the kick to sound compressed.

Maximizing a full track you usually dont want it to sound compressed, usually more transparent.

I will be glad if you will fix me.

Let me actually recommend that you try this:

On either your kick or bass track - set a high pass filter for around 30Hz with a q about 1 (if you're using SOntius, as a reference). Different plugins have different "Q" settings ;unfortunately.

ALso - you can, as I do, create default "normal.cwt" (template that you start with) that has this rolloff already there; then I adjust to taste.

As simple as this may sound - try it. The fact is - listen real carefully to "pro" stuff and you'll see it is more "perceived" bottom and loudness. If you listen critically, you'll realize things are actually SMALLER as the whole audio spectrum in a given project needs to fit into "speakers", which are NOT the same as the human ear.

And the funny thing is - if you roll off at least at 30Hz, you actually can pump more 100Hz (for example) because the rollofff works almost like a compressor ... because it removes the extraneous boomy stuff and lets you "manage" the lower frequencies better.

Check it out, see what I mean. And it's a pretty common "default" way to work with sound/audio (that 30Hz and down stuff, that is). This kind of thing has more to do with the science of audio than "opinion', so to speak.

ORIGINAL: Plyrman

Hey guys and gals.....

I'm doin Jazz and R&B....and the mix down process is driving me nuts......I can basically get everything where I want it except for the "kick and bass" ....the two most important elements......I purchased the Mixing Engineers Handbook, which helps ..but still the kick and bass are not where they should be....

I guess I'm searchin for some "actual" settings....The "Handbook" just gives you the "frequencies"...but not actual settings....

The kick and bass are either too loud or too low...meaning I get a rumble instead of hearing the instruments clearly....

At present...I've got the kick bottom around 80 and cuting 8db....the top at 2K and cutting 2db...I get a little confused with the two bands in the middle...so nothing there....

the bass settings are..bottom....130 cutting 4db ...800 boost of 4db and the top at 5k cutting around 2db....

when I play the mix....I have the levels down but after burning to CD and listening...they still sound "Boomy"....??? If anyone has any tips ...well THANKS very much in advance.....

ORIGINAL: bantry52

Hi guys!! I've just started using the Cakewalk software and I'm having a problem when I try to burn a CD. Somehow the last 20 seconds of the recording is being cut off. Is this a problem in the burning process or in the recording process? Has anyone had a similar problem? If so, how would I correct this?

Any suggestions would be greatly appreciated.

Hi, welcome!

Best to start a new topic thread rather than tag the question on here, it'll get lost. Try posting it in the relevant forum for your software.

I'm telling you again. YOU NEED TO PUBLISH A BOOK ON MIXING.

I'd buy your book Yep, based on everything you've posted on this forum - thanks!

@ Yep


You sir, are a GOD

but he's not posted in this thread since 9/11/2008. someone resurrected this old thread - which is very good - but yep's not been on these forums much at all since last fall. he's not likely to see your props.

ORIGINAL: Beagle

but he's not posted in this thread since 9/11/2008. someone resurrected this old thread - which is very good - but yep's not been on these forums much at all since last fall. he's not likely to see your props.

First, pull down all the faders. We're going to start over completely from scratch. Turn up the kick and snare and get them so they're hitting about -6dB. Don't worry about anything but the levels right now, just get those two instruments sounding good together. One of the reasons it's hard to provide "settings" is because every instrument and recording is different. So I'm going to show you how to make your own "settings" for your track.

Take your equalizer and dial in a boost of around +12dB with a sharp Q (maybe 2 or so). Now, start to sweep that through the frequencies and listen to the effect that it has on the kick. You are looking to identify three specific components of the sound: The dull "thump" (probably somewhere between 60-120 Hz), the "click" of the beater head (probably somewhere in the upper midrange, 2-6kHz or so), and the round, boomy "note" of the resonating drum (probably between 100-250 Hz). As soon as you find any of these "sweet spots," stop. Leave your eq node "parked" at that spot as a marker, and bypass that frequency or turn it down to flat. We are not going to DO anything with these frequencies right now, we just want to identify where they are, because they will be useful later.

Wherever you find the "thump" of the kick drum is the lowest useful frequency, so take a highpass filter (low cut) and roll off everything below that "thump" frequency. If you feel that's too drastic or you want to leave something for the subwoofer fiends, use a shelving filter instead and turn it down by 6dB or so. This is going to clean up rumble and subsonics that will eat up headroom and waste your time.

Next bring up the bass and get that to fit in good with the kick and snare pattern. Use the same technique as above to roll off the ultralows, and also roll off the extreme highs of the bass, say above 10k or so-- the only stuff that lives up there is hiss and fizz. Now to identify the useful range of the bass. Unless your bass player plays slap style (please tell me she doesn't), the bass is likely to need one full octave and only one octave to really be the bass. so we're going to look for one octave of the frequency range that best showcases the stuff that is really critical to the bass.

Bring up a high shelf and a low shelf filter on your equalizer and dial them down 12dB or so each. Now drag the frequency points so they turn down everything but the octave from say 100Hz to 200Hz (An octave=a doubling in frequency). Now bypass the equalizer and A/B that with the bandwidth-limited sound. Do you need more lows? if so, drag the filters down to see how low you must go. If you need to go higher, go higher. Just remember we are hoping to identify one octave, so if you go up to 300Hz, try and cut the lows at 150Hz and so on. You don't have to be religious about it or anything, you just want to really try to narrow your focus and decide what's important.

Once you have identified a range of about an octave (you can go a little wider if you have to), bypass the eq but keep the nodes in place. Again, we're not USING these eqs to modify the sound right now, we're just establishing "markers" that tell us where the boundaries are. Those boundaries mark out sacred ground that the bass must have full ownership of. When you start to dial in other instruments, they are going to "mask" parts of the bass sound, and that's fine. They may cover up the string articulation and that nice woody tone and all that and it's sad but it's okay, because other instruments do mids and highs better than the bass does. But none of them should mask this range of frequencies. They might share it a little bit if the bass doesn't mind, they might overlap it a little, but the bass must dominate this octave. If another instrument is causing problems, eq this range out of that instrument.

So now we know where the important parts are of the kick and the bass. Start to bring up your other instruments, vocals first, and get THOSE instruments to sit in the mix with the bass and drums, not the other way around. If the kick drum starts to sound too boomy, you just turn down the "note" frequency that you marked earlier. If it needs more "thump", you just put a couple dB boost on the magic button. You can make little changes that are so small you don't even hear them as a change, they just make things seem to fit better and "feel" right. And remember, the eq settings we used above were just as a tool to FIND those places, they are not the settings we use to sculpt the sound. If I wanted a little more "click" to help the kick cut through the mix I'd prolly start with about 3dB boost Q around 1-1.3, whatever sounds good. Make sure to make these adjustments with the whole mix playing.

Oh, and turn down your speakers. Way down. Mix quiet, like below conversation level. It will help you to keep everything balanced and in perspective. And take frequent breaks.

Next chapter is on compression.

Cheers.

_____________________________

Boycott Monster Cable
companies being sued by Monster
example of despicable business practices

Better said than any book out there... And I've read a lot!

keep on reading...
http://forum.cockos.com/showthread.php?t=29283&highlight=mixing

ORIGINAL: Dave King

Looks like Yep's a Reaper user now. Big loss for Sonar community

...which begs for an answer to the question: Why?

Nothing sinister!

I still use and own cakewalk products.