Part 1 of this series focused on Compressors, Limiters, and EQ. Part 2 explores Noise Gates, Delay, and Reverb.
In addition to your microphones, DAW/console, and room, an essential part of any home studio set-up is your signal processing gear. From the dynamics control of compressors and limiters to the effects processing of reverb and delay, these tools are necessary to create a professional-sounding final product. But for the new engineer, these effects can be fairly mysterious, and a tendency to overuse plug-ins and outboard gear is commonplace, especially for someone just learning the nuances of the art of recording.
How can you best use your typical signal processing plug-ins to enhance and optimize your recording? Understanding how the dynamic control processors like compressors, limiters, EQs, and gates function, and knowing how to use multi-effects such as delays and reverbs to perfection will make you a better producer and engineer. It’s also important to remember that signal processing tools are just that – tools. There are no rules stating you can’t use them in different or novel ways to create new sounds. But before doing that, it makes sense to learn about the basic parameters of each and the functions they were invented to serve.
The floor tom was close mic’d, and now, listening back critically, the drum’s nearly five-second release time blurs the tom’s definition. You can live with some of the release but you want to clearly hear the attack of each hit on the floor tom. The best signal processor to help solve this is a noise gate. Noise gates are part of the dynamic processing family of plug-ins. Like compressors and limiters, the noise gate has a user-definable threshold, provides variable gain reduction, and offers attack, hold, and release time parameters.
Gates function by setting a threshold level that determines the amount of amplitude required to open the gate, then letting the audio pass through to the gate’s output. Any amplitude level below the threshold value will not open the gate – so the gated track remains silent. On this particular song, there are a few breaks that Doug leads into with the floor tom, but it rings on too long. Loop the phrase so it plays continuously, then insert the noise gate on the floor tom track and set the threshold level to the point at which the hit on the tom just barely opens the gate. Now adjust the attack, hold, and release parameters to achieve the desired floor tom effect reducing the long decay.
Noise gates are very useful when you need to eliminate any unwanted incidental sounds that may have been recorded. For instance, use one on vocals to eliminate breathing sounds between lyrical phrases, or on a distorted lead guitar to eliminate overdrive noise between lead passages. Noise gates could even be tried on the stereo mix bus output to really tighten the breaks in the song.
Noise gates can also create their own problems, since everything recorded on the track you are gating is eliminated according to the gate’s envelope, including any ambient leakage. This can sometimes cause a perceptible and distracting dropout within the song’s mix. As with all signal processing, use your own ears to decide how much noise gating is useful in your mix.
Delay and Reverb
So far the plug-ins mentioned in this article are generally considered to be dynamic, having to do with varying or controlling amplitude in real-time. The next type of signal processing is time-based processing, and we’ll focus on the bread and butter effects of delay and reverb.
In our example, the bass player recorded her parts directly into the DAW interface via direct box. Her Fender Precision Bass sounded great, and with a little EQ and compression, the track is all set. The guitar player tracked the leads with his guitar processing pedals, but recorded the rhythm guitars direct and dry. Now you are faced with the challenge of giving life to his rhythm guitar parts.
Let’s start with delay. A delay is a time-based processor that generates discrete wave fronts of the input signal according to the delay time. Delay settings of 250 to 500ms will create rhythmic interest while smaller times such as 20 to 80ms can create a sense of depth. You can also create echo effects by increasing the amount of feedback, a parameter that returns the output of the delay circuit back into itself.
Many delays provide rhythmic note values, such as whole, half, quarter, eighths, etc., and offer a sync option that times the delay precisely to the tempo of the original track. The delay also has low and high-cut filter parameters, so you can change the frequency content of the delay generation when feedback is used. You can also modulate the delay time using the depth and rate parameters, and create variable moving rhythmic echoes.
Here’s one practical approach, assuming there are two rhythm guitar tracks. Start by bussing one rhythm guitar to an Aux Track and insert a medium delay. Set the delay time to 40ms and pan the Aux Track to the right, leaving the original rhythm guitar in the left channel, creating a delayed stereo spatial spread.
For the second rhythm guitar track, a long stereo delay provides a good option. For this plug-in, a stereo Aux Track is required, or if inserting on a mono audio track, you can automatically convert it to a stereo track. For the most part, the controls are the same as the first delay used, but now there are separate left and right channel parameters on the delay itself, allowing you to create complex rhythmic and spatial movement in the stereo field.
Finally, you turn to the last solo instrument, a melodica, a three and a half octave reed instrument played by blowing into its mouthpiece and fingering its keys. It’s a warm sounding instrument that requires some compression but generally sounds fine. Here the decision is to add Reverb.
Reverb (short for reverberation) is one of the oldest and most widely-used time-based effects. It can add lush ambient room sound to any instrument. Like delays, reverbs generate multiple wave fronts, but there are a large number of fronts and the time differential between each front is extremely short. It’s easiest to think of these fronts as reflections of the original sound, like the way an instrument sounds when played in a well-designed concert hall. The sound generated by the instrument moves out in all directions. It comes directly toward the listener but it also hits the floor, walls and ceiling. The sound reflections from these surfaces return back to the listener slightly delayed from the original sound, depending on the size and depth of the space. Of course the reflections off the floor, walls and ceiling also continue to bounce off of the surfaces in the space and listeners perceive all those reflections at slightly different times, creating the perception of a spacious concert hall.
Today’s reverbs emulate a wide variety of acoustical spaces. Some of the most common environments include Hall, Room, Church, Club, and Stage. Some reverb plug-ins offer additional emulations taken from the analog reverb days such as Plate, Spring, and Chamber. In all cases there are a few common parameters that can be selected and adjusted to good effect.
Reverb type refers to the room being emulated (hall, room, arena, church). Reverb size refers to how large of a space you can create. You might have a large room, a small church, or a medium hall. Diffusion is a parameter that determines how far apart each reflection spreads out from the instrument, giving a sense of depth of the enclosure. Reverb Decay adjusts how fast the reflections die out after the initial attack of the sound. Pre-delay is an important parameter that determines the time differential between the direct sound and the point at which listeners perceive the reverb reflections. Finally, most reverbs have low and high cut filters that can reduce or increase harmonic partials as a part of the reverb’s reflections. These filters are very useful to create transparency within the reverb process.
It is important to remember, the best sounding reverb is the one that enhances the sound without being too noticeable. For the melodica solo, the large hall setting, a pre-delay of 40ms, a wide diffusion and cutting high frequencies at 8kHz, results in a dreamy-sounding solo for this tune. Everything is now set to begin the final mix, with signal processing tools helping to address the issues that would have made this EP project sound less polished.
When using plug-in processing it is critical to keep in mind the style of project on which you are working, the type of instruments you will be recording, how they will be recorded, and what kind of plug-in processing will help when it comes time to mix. As you get more familiar with how signal processors work, listen to some of your favorite recordings and try to reverse engineer what types of processors and settings may have been used.
Addendum: The Lowdown on Impulse Reverbs
The various reverbs known as impulse reverbs offer a different approach to time domain processing. These plug-in effects processors provide reverberation, but their processing method is very different from the traditional reverb plug-in. Impulse reverbs, although they have many of the same parameters mentioned above, access a complete library of sampled sonic spaces known as impulses, taken from various rooms known for their unique acoustical characteristics found all over the world.
Famous concert halls, cathedrals, and classic recording studio tracking rooms are just some of the options available when shopping for impulse reverbs. Instead of algorithms that emulate or calculate the dimensions of a hall, church, room, the impulse reverb actually loads the acoustic signature of a given space with all the actual time variables included. This results in a totally convincing audio reverberant spatial environment.
Addendum: Using Auxiliary Tracks to Preserve Your Computer’s CPU Power
It should be noted that reverb and delay plug-in processing demands more CPU power and larger amounts of RAM in order to accommodate the time differential between the input of the unprocessed signal and the output of the plug-in processor. Because of this time differential, something known as latency (the time difference between the audio signal written to the drive, passing through the CPU, getting processed, and then returning to the audio output) can result in slight phase anomalies. Most professional audio recording applications compensate for latency by imperceptibly delaying the returning audio to the track output. The user is never aware of the delay and for the most part operates the application as usual. However, it is important to keep in mind the total number of time-based processors you insert on any given audio track.
The more time-based plug-ins inserted on a track, the greater the amount of compensation required. For this reason, when using time-based processing, you may wish to create an Auxiliary track (Aux Track) as you are setting up your mix and insert the time-based processors on the Aux Track and then bus the audio via the individual channel sends to the Aux Track. You can set up one for reverb, another for delay and so forth. In this way, the original audio track with the recorded instrument information is separate from all the time-based plug-in processors, minimizing the need for latency compensation of the audio track. Doing so will result in a sharper and more defined audio image throughout your mix.
Recording with Reverb and Echo – Tips and Lessons from Six Classic Tracks (November 2010)
Keith Hatschek is a regular contributor to Echoes, author of two books on the music industry and directs the Music Management program at University of the Pacific. Jeff Crawford is a recording engineer and producer with more than 30 years industry experience. He also teaches music technology at Pacific.