A common technique in modern music production is the practice of layering multiple drums sounds or samples to achieve a specific sonic goal or texture. For example, taking the sustain and bottom of a nice 808 kick, the fuzziness or grit of an acoustic drum, and layering the top end “snap” of an electronic kick sample can be an affective way of achieving just the right blend of sonic texture to cut through the mix and compliment the rhythm section, many times creating complex layers that could not be achieved with basic EQ or Compression of a single source alone.
However, this increased flexibility often comes at the cost of bandwidth clutter. One of the biggest problems with combining low frequency samples, especially kick drums, is the accumulation of mud and rumble, along with phase coherency issues that can end up helping or hurting your cause. Because of this, you need to be clever in how you choose to blend your sounds, sometimes by removing unnecessary frequencies or altering competing envelopes, fitting each of the pieces together like a jigsaw puzzle to achieve the final combined layer.
In my latest Lynda.com course, Mixing a Hip Hop & RnB song in Pro Tools, I discuss the practice of layering kick drum samples to create compelling textures, as well as techniques to ensure sonic compatibility inside your mix. Check out the following free video from the course, be sure to switch over to the HQ stream on youtube and listen on speakers or headphones with a decent amount of bass extension, as I’m working with very low sub frequencies. If you like tutorials and tips like this, be sure to unlock access to the entire course and all my courses over at lynda.com
The speed at which a signal comes and goes as the compressor grabs and releases the program material is controlled by the processor’s attack and release controls. While a compressor’s envelope can be used to shape a signal in creative ways (e.g. make a snare drum punchy or a vocal extra breathy) it can also work against us if used improperly. When first learning to use compressors and limiters, it can be difficult to hear the compressor’s envelope, or attack and release settings, making them difficult if not impossible to set correctly by ear. While there is a certain amount of practice and experience involved with hearing the subtleties of attack and release, especially at low levels of gain reduction, one thing that I find that helps me hear these parameters working is to monitor at lower listening levels.
Our monitoring equipment (speakers and amplifiers), as well as our human auditory system, begin to naturally compress material at very high SPLs (very loud listening levels). While monitoring your mix at extreme levels is not a great idea for a variety of reasons, namely fatigue and hearing loss, it is especially difficult to hear subtle envelopes in your compressor’s attack and release settings when so much gain is competing for your speaker’s (and ear’s) attention. In other words, if your speakers are so loud that your ears and amplifiers are also adding compression to the signal, how can you expect to make critical decisions about compression?
This is not to be confused with discussions over equal loudness contour graphs and the frequency response of our hearing at different SPLs (there is a whole explanation/discussion of that in my Foundations of Audio: EQ and Filters course ). I am talking about the ability to accurately hear subtle gain changes in a signal over fractions of a second. While I might crank up the volume when EQing my bass and kick drum, to take advantage of my ear’s flatter frequency response at higher SPLs, I will work at much lower levels while listening for subtle changes in a signal’s envelope during compression.
Another trick I find useful when trying to get a sense of my compressor’s timing is to initially use deeper threshold and ratio settings, then ease back when I find something that works. Working the compressor harder creates more push and pull and can help you hear the envelope of the gain reduction, because a greater gain change is happening during the attack and release phases.
It is important to note that compressors attack and release cycles (while often labeled in milliseconds) can be radically different from one another, even varying with the type of signal and the amount of gain reduction being performed. In other words, one compressor plug-in with an attack of 10ms may react entirely differently than another compressor set for a 10ms attack, and the same compressor’s envelope may react differently to one instrument versus another. This is why I suggest that instead of trying to acquire and learn every compressor on the planet, you set aside a few “go-to” processors to learn extremely well. This way, you will be able to predict how they will react under different mixing scenarios, allowing you to translate the sound in your head faster.
At the end of the day, manipulating a signal’s envelope using a compressor’s attack and release controls is an entirely subjective task, with one man’s punch and swagger sounding like another man’s unwanted pumping and breathing. The sooner you can clearly hear the envelope working, the sooner you will be in control of your own esthetic, and form your own opinions about which compressors you prefer on specific material.
To learn more about attack and release, as well as other compression and limiting techniques, be sure to check out my course Foundations of Audio: Compressors and Dynamic Processing @ lynda.com
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Understanding the fundamental building blocks of sound is an important step for any mixer, producer, or musician. These core concepts are ultimately what bring our speaker cones to life, sending the sweet sounds of our favorite music through the air and into our ear canals. Check out this free video clip from my course Foundations of Audio: EQ and Filters, be sure to watch the whole course over at Lynda.com.
This entry was written by Articles, MixTips, News, video and tagged EQ, equalizers, mixing, recording, sound. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
A “pass” filter affects all energy above or below the target frequency, often referred to as the “cut-off frequency,” by removing that frequency content in a continuously decreasing downward slope. Pass filters are extremely valuable tools in mixing because they allow us to”bracket” the frequency energy of a given signal in the mix. We can use hi-pass filters to remove low-frequency mud or rumble from the recording process, or lo-pass filters to shave off hi-frequency content, pulling the signal back in the mix.
While most intermediate mixers are familiar with using hi-pass filters to remove mud and clear out unwanted frequency content in the low-end of a mix, many are unaware at how valuable low pass filters can be at smoothing out your mix’s frequency response and creating clarity and contrast amongst tracks. Unlike an analog mixer or analog tape, a DAW’s mixer is 100% frequency linear in it’s summing process, meaning that no frequencies are altered or “soaked up in the circuitry” during the summing process. Combine this with the fact that many modern virtual instruments and loop collections have an insane amount of high frequency extension (have you heard some of the “paint peeling” top end of plug-in synth presets lately?) and a mix can gather a boat load of fatiguing top end very quickly.
Aside from creating context (or lack there of) issues between the focal tracks in a mix (e.g. if everything has a ton of top end, the mix can lack a sense of depth of focus on the elements that you want to have pop out), I find that this build up of ultra high frequency content tends to do a few nasty things. First, all those high frequencies, if left un-checked, ram into the D/A’s anti aliasing filter all at the same time, and depending the quality of your D/A converter, can result in a nasty haze in the top end that muddles the stereo image and becomes very fatiguing to listen to over time. Second, and most important for a lot of the commercial work I am doing, I find that feeding a mix with way too much un-checked top end into mp3 (or any other compression) algorithms tends to force the algorithm to sacrifice important mid band bits in an attempt to retain a top end you can barely hear. In other words, mixes that aren’t insanely bright will tend to fair better under heavy audio compression.
Now I am not saying go and filter every track in your mix down to 10K, but I do find that subtle lo-pass filtering at very high frequencies with transparent EQ filters (e.g. filtering 6dB/oct at 12K, 15K, 20K, etc on certain non-focal tracks) can help give me a more analog feel and ensure that my mix sounds better when run through youtube’s compression algorithm 50 times.
To learn more about hi and low pass filters, check out this free clip from my Foundations of Audio: EQ and Filters course or watch the whole course at here at lynda.com
This entry was written by Articles, Audio and Video, MixTips, News, video and tagged EQ, equalization, mixing, producing, recording, song writing. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
As humans, we have tons of built in “thresholds” surrounding all of our senses. What do you do when the shower gets too hot? As the temperature rises, it eventually gets to the point that your nerves tell your brain “ouch” and your brain tells you “too hot!” and you turn down the hot water. That point of “too hot,” be it 110 degrees or 120 degrees, is your skin’s personal threshold for water temperature. In the case of a compressor, the threshold control defines the point at which the compressor will begin to alter the dynamics of the signal, or compress. If the signal breaches the threshold or gets “too loud,” the compressor reacts by compressing or “turning down” the level of the output, just like you turning down the water temperature in the shower. In other words, It’s the threshold control that tells the compressor when to compress and when to leave the signal alone.
All dynamic processors work on the simple principle of a defined action resulting in a prescribed reaction. Think of it as an “if..then” statement of sorts; if the incoming signal level reaches “X” or higher, the processor will react in “Y” way, if the signal remains below X then do nothing. The threshold control in a dynamic processor allows us to define the reaction point, or the “if” part of the statement, setting the level at which the processor reacts, sometimes by reducing (compressing) or increasing (expanding) the level of the output.
I know what you’re thinking, you’ve seen that threshold control just sitting there, but the presets you pick automatically set for you, so what’s the big deal? Indeed, most compressor plug-in presets include a preset threshold value, and while it might be tempting to use that level, it is very important to understand that the preset threshold may not be affective for your specific compression task. If the threshold is set too deep, it might over compress the signal, removing all the dynamics. If the threshold is set too light, the compressor might not engage at all. Of course, sometimes you get lucky and everything works out, but remember, the preset has no way of knowing what your input level looks like or how much compression your signal needs. Until we have intelligent processors that can adjust their threshold automatically to accommodate the input signal, you will need to adjust the threshold control manually. Even if the processor could detect and adjust the threshold automatically based on the input signal, the preset couldn’t know how much compression you needed or desired, just like no one else would know just how hot you like your showers on any given day.
Ultimately, whether you set up your dynamic processors from scratch or from a preset, it is imperative that you understand and use the the threshold control to achieve the desired amount of processing. If you want to learn more and see some cool animations, watch the following free clip on understanding threshold from my new course lynda.com, Foundations of Audio: Compression and Dynamic Processing. Be sure to check out the entire course for more tips on using threshold and other compression controls.
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When you use traditional compressors to alter a signal’s envelope shape, you often trade one problem for another. Spiking out the attack uses deep threshold settings that alter the tone and shape of the decay, sometimes in an undesirable way, forcing us to use “mults” or parallel chains to achieve the perfect attack and sustain characteristics for a given instrument. Transient shapers allow us to modify the transient and sustain portions of a signal’s envelope in very transparent and discrete ways, without altering the tonal character of the signal. To learn more about this topic, check out the following free video from my new lynda.com course Foundations of Audio: Compression and Dynamic Processing.
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Dynamics processors are little more than automatic volume controls, responding to changes in signal level over time. If the input signal’s level meets a specific criteria (threshold) then the processor reacts by changing the output signal level in a prescribed way (turns it down, turns it up, etc). Here’s another clip from my new Lynda.com title Foundations of Audio: Compression and Dynamic Processing. Check it out.
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Fresh from my brand new Lynda.com course Foundations of Audio: Compressors and Dynamics Processors, here is a free video on gating a drum track. You can watch all the videos in this new course by signing up for a free 7 day trial.
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Just because you failed high school trigonometry doesn’t mean you can’t use and profit from parallel compression in your mixes. Parallel or “upwards” compression is simply the process of combining an un-compressed signal with a compressed-one and blending to taste. This ‘best of both worlds’ approach is designed to preserve the dynamics, openness, punch, character and frequency response of the un-processed signal while solving the issue of the overly dynamic track getting lost in the mix or sound thin/weak.
While many newer dynamics plug-ins feature a built in “wet/dry” mix parameter that allows inline parallel compression tricks, you can easily achieve this effect with any compressor/limiter. Basically we are going to duplicate the track we want to compress, add a lot of compression to the duplicate and blend with the original to taste. Essentially what this is doing is creating a dense ‘bed’ of sorts for the uncompressed track to ride on, preserving all the original dynamics while allowing the track to sit comfortably in the mix. This trick can be used in subtle or extreme ways and works well on almost any source material, especially where transparent and natural sounding dynamics control is desired.
Parallel Compression Tips:
Drum Squash: Create a aux return with an aggressive compressor or limiter on its insert, you can call this “Squash Bus.” Using sends, send all your drum tracks accept the kick drum to this squash track and blend with the original (dry) drums to taste. Leaving the kick drum out prevents the squash track from over-reacting to the dynamics of the kick, which tends to dominate the other drums. You can experiment with including the kick to create cool pumping effects on the squash track. Try EQing this squash track in different ways, or even add distortion for an over-the-top effect.
Automation: Automate your parallel track up and down at different sections of the song, bring it in during the choruses for more power and support, bring it down during the verses for a more intimate feel.
Using a Limiter: For natural dynamics control on vocals, guitars, etc. try using a very fast (brickwall style) limiter on the uncompressed track just to keep the peaks under control so nothing jumps out of the speakers (you know, in that uncontrolled, “karaoke” sounding way). Then use the parallel track to subtly bring up the valleys and fill in the body/sustain of the track. This works really well on vocals with a ton of plosives (e.g. hard P or T sounds) that you don’t want to over compress. The brick wall limiter will transparently grab and control those plosives transients while the parallel track will bring up and support any softer sections without having to squash the crap out of the vocal.
Compensate for any delay to maintain phase coherency: Because some compressors/limiters incur a small amount of processing delay (usually do to look-ahead algorithms) it is important that each component of the parallel chain is delayed by the same amount. For instance, if you were to create a duplicate track of a vocal and apply a L1 Maximizer to the duplicate (parallel) track, there would be a noticeable latency and serious comb filtering would be heard when both tracks are played together. Most DAWs (except for Pro Tools) handle this automatically as part of their built in PDC (plug-in delay compensation) engine. In Pro Tools LE/Mpowered, the easiest way to solve this is to copy the plug-in to the original “dry” track and bypass it, thereby incurring the same delay on each track. A more permanent solution would be to shift the duplicate track backwards by the amount of delay the plug-in is causing. In Pro Tools HD you should always use delay compensation when mixing/editing. Remember, not all plug-ins incur a delay, in fact many do not, so considering using those when creating parallel chains.
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So I use quite a few “wrapped” RTAS plug-ins, meaning that the plug-in was designed for VST and hacked (via FXpansion’s VST-RTAS wrapper utility) to work as RTAS in Pro Tools. For example, all of the UAD stuff shows up as wrapped plug-ins in my inserts menu. This isn’t a big deal, as with most VST plug-ins the wrapper utility works flawlessly, but no matter which plug-in you end up using, it hides itself in the “Wrapped Plug-ins” sub-menu of your inserts menu. I have talked to Avid (Digidesign) about this and unfortunately, as long as they are “wrapped” VST plug-ins there is no way to get them into their correct categories (e.g. EQ, Dynamics, etc), aside from possibly hacking the plug-in .dpm file (I’ve looked into this. Without some sort of hint as to how the category identifier is determined/stored, I can’t see anything obvious when I pull up the files in a hex editor). While there is currently no easy way to get the wrapped plug-ins into their correct catagories, there is an easy workaround of sorts that will get you to your favorite plug-ins quickly.
Default EQ and Dynamics:
The first thing you want to do is set up your mixer’s default EQ and Dynamics plug-ins, these will show up at the highest level of the insert selection menu so put your “go to” EQ and compressor plug-ins in these two slots. To set the default EQ/Dynamics choose Setup > Preferences > Mixing Tab.
The default EQ and Dynamics option only allows you to save one favorite within the EQ and Dynamics sub-menu, which blows if you want use a wrapped plug-in as your favorite EQ or compressor. In this case, you can save a plug-in as a “favorite” and it will show up at the top of the plug-ins list (within the plug-in type sub-menu on stereo tracks, e.g. TDM/RTAS or multi-channel/multi-mono).
To save a plug-in as a favorite: Hold Command (Mac) or Control (Windows) while selecting a plug-in from the inserts menu. The plug-in will not be inserted but will be stored as a favorite. Remember, you must hold Command (Mac) or Control (Windows) before clicking on the track’s insert selector.
To remove a plug-in favorite: Repeat the steps above, hold Command (Mac) or Control (Windows) and re-select the plug-in stored as a favorite from the insert selector menu.
You can have as many favorites as you’d like however keeping this list small will ultimately save you more time.
This entry was written by Articles, MixTips and tagged mixing, MixTips, plug-ins, pro tools. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.