AF’s Weblog

June 1, 2012

How to Get the Pumping Drums Effect with Sidechain Compression

To read the full article see:  Sidechain Compression

Sidechaining has been around for years; this is the process of using one signal to control another. A couple classic examples are using a kick drum to gate a bass part, or doing de-essing – isolating the high sibilant frequencies from a vocal, and using those to trigger compression so that the sibilants come down in volume.

But in the digital age, we can do a lot more with sidechaining. One of the most popular applications is with dance music, where sidechaining can create the “heavy pumping” electronica drum sound used by artists like Eric Prydz and others.

We’ll describe how to do this with Cakewalk Sonar, although the same principle applies to other programs that allow for sidechaining. Sonar allows sidechaining for several effects, including compression, so that one instrument can control the compression characteristics of another instrument. This offers a variety of effects, including a “pumping” drum sound for multitracked drum parts; we’ll do that by setting up the snare to control compression for all drum tracks.

Fig. 1: You’ll need a drum submix bus to create an overall drum sound.

The first step is to create a drum submix bus, and send the drum tracks to it (Fig. 1). We need this submix so the entire drum track can be processed by the sidechained compressor. To create the submix bus, right-click in an empty space in the bus pane and select “Insert Stereo Bus.” To create a send in track view, right-click in a blank space in the track title bar and select “Insert Send.” From the menu that appears, select the send destination. Make sure you feed the bus pre-fader, and turn the individual drum channel faders down so that only the bus contributes the drum sound to the master.

Fig. 2: Assign the Drum Submix out to your main stereo output.

Let’s take a closer look…

….

Create a second pre-fader send in the snare track, and assign its out to the bus feeding the sidechain input.

Fig. 7: We’re almost there – it’s time to adjust the compressor.

To adjust the compressor, start with the compression attack time set to 0 ms; the drum sound will essentially disappear when the snare hits because the gain is being reduced so much. Gradually increase the attack time to let through more of the initial snare hit, and add a fair amount of release (250-500 ms) to increase the apparent amount of pumping.

And there you have it – the pumping drum sound. May it go over well on the dance floor!

To read the full article see:  Sidechain Compression

Advertisements

April 20, 2012

Mixing Rap Vocals – Part 3: Compression

Filed under: Compressors, Mixing reviews — Tags: , , , , , — audiofanzine @ 8:54 am

To read the full detailed article see:  Tips for Mixing Rap Vocals: Compression

Time for the third installment of the Mixing Rap Vocals series: Compression.

I highly recommend you check out part 1 & part 2 before reading this article.

Compression is a difficult subject because there is a lot you can do with it. So let’s look at the main reasons to grab a compressor before getting into some of the more intricate uses.

Quick Macro-Dynamic Control

Macro dynamics refer to words and phrases. These are the clear dynamics you can hear as “this part is louder, that part is softer.” The most transparent way to get things sounding even is to actually automate the vocals manually. But sometimes time doesn’t allow for this approach. So if you aren’t automating, a light ratio, slow attack, slow release, just catching the louder moments with the threshold is a good way to even things out.

Micro-Dynamic Control

What volume automation might not catch is the very quick dynamic changes – loose spikes at the fronts of words. These spikes aren’t heard so much as “volume” but more as an overall quality to the vocal.

The issue with these spikes is two fold – first, they eat away at your headroom pretty quickly– second, they will trigger any compressors you are trying to use for purposes besides micro-dynamic control.

It can be useful to dedicate a compression stage toward pulling back these vocal spikes. Generally a fast attack and release, and a light ratio does the job. The light ratio is to retain the articulation of the word and minimize frequency skewing. The key is to set the threshold low enough to catch as much of the peak as possible while effecting the body of the signal as little as possible. I try to avoid using limiters for this purpose. I like the Empirical Labs Distressor for this (especially for controlling peaks while tracking), as well as digital style compressors such as the Logic or Pro Tools stock compressors or the Waves C1. The attack setting is very important – it’s usually between a number of nano-seconds and two or three milliseconds in the digital world, and on the faster side of things for the analog world (totally varies unit to unit).

Getting a Vocal to Stay Audible Through a Mix

The power of compression is that you can make something louder while not actually raising the peak volume of the signal. This becomes extremely useful for making something cut through a dense mix or to come forward. This is probably where the majority of compression work for rap vocals come in.

Rap is generally an in-your-face, visceral style of music. The kick is physical, the snare is physical, subtlety isn’t really the overall goal. And the vocals are paramount. I’ve mixed a number of rap records where the vocals are lower in the mix, but never have I thought it was a good idea. Generally I want the vocals to be equally as strong as the drums or stronger, and I want them as “forward” as possible. Compression is usually a part of that equation.

Let’s consider some more issues…

Conclusion

Compression is a powerful tool that many people struggle to fully understand, so try to get your hands on one and start experimenting. As always I’ll keep an eye on the comments in case there is anything that needs clearing up. I also encourage you to share your own compression tips!

To read the full detailed article see:  Tips for Mixing Rap Vocals: Compression

March 12, 2012

Tips for Effective Buss Compression

Filed under: Compressors, Mixing reviews — Tags: , , , — audiofanzine @ 2:31 pm

Buss compression is certainly not a new concept, however, it is an effective and reliable engineering tool and its basic principles are vital considering you are affecting multiple voices.

When approaching buss compression, there are two essential tools at your fingertips: Attack and Release – these two tools, when properly utilized, will have the ultimate say in the outcome of your efforts.

The attack and release functions of a compressor will tell its detector how to react to signal that passes through. An effective use of attack and release will essentially allow you to make conscious envelope changes to the signal rising above the threshold at the detector. This brings about the main philosophical concept behind compression, which is to shape the signal, rather than merely restrict its dynamic range (dynamic restriction is part of shaping the signal, not the end purpose). The attack and release controls are what really provide the push and pull effects of compression.

With this in mind, I have provided examples of effective and ineffective buss compression, focusing on attack and release settings, for a few simple approaches.

All of the following audio passed through the same compressor with the same settings (beside attack and release) and a ratio of 1.5:1 with an average gain reduction of 4 dB.

To read the full article with sound samples visit:  Buss Compression

October 4, 2011

IK Multimedia T-RackS Black 76 & T-RackS White 2A and Native Instruments VC 76 & VC 2A Review

The (almost) simultaneous launch of the 1176 and LA2A software versions by IK Multimedia and Native Instruments is a good opportunity to make a quick comparison. Let’s go!

Some hardware products —instruments, signal and effect processors— have a kind of Holy Grail status. Among studio processors —and regardless of their denomination: limiting amplifiers, leveling amplifiers or just compressors— the Teletronix LA-2A and Urei/Universal Audio 1176 LN, as well as the legendary Fairchild 660 & 670, which are extremely rare to find (we saw a unit sold for $42,000 on ebay…), are highly regarded pieces of gear you’ll still find in big studios either as original or reissue versions. Many recording studios also use more or less faithful replicas of the originals designed by manufacturers ranging from Studio Electronics to Purple Action that (mostly) have a great sound quality. It’s simple: these legendary tools can be heard on almost every album ever produced since they were first introduced.

As expected, the software world also has its own interpretation of these legends. Since the early attempts by Bomb Factory to the latest products by IK Multimedia and Native Instruments in collaboration with Softube, and including the existing Universal Audio, URS and Waves products, the market is packed with software simulations.

We do not intend to review all software versions (they are too many) nor to compare them with vintage or modern hardware products. We just want to compare two manufacturers and use the same plugins for Universal Audio’s platform UAD-1/2 as a reference, without neglecting the performance of the original processors and some hardware replicas. Note that Native Instruments’ Vintage Compressors bundle also includes the famous dbx 160 compressor, which we won’t take into account for this review.

Introducing the Plug-ins

Test system

MacPro Xeon 3.2 GHz

OS 10.6.7

Logic 9.1.4

IK Multimedia T-RackS3 Black 76 and White 2A v.3.5.1

Native Instrument Vintage Compressors VC 76 and VC 2A

UAD-2 UAD 1176 LN and UAD LA-2A v.5.9.1

Native Instruments decided to collaborate with Softube, a manufacturer that has launched quite exceptional plugins — I can’t seem to get enough of — like the Acoustic Feedback and the Tubetech CL 1B. I haven’t had the opportunity to try out their reverbs and amp simulations yet, but other user’s feedback is very promising. So let’s start with the Urei 1176 LN and Teletronix LA-2A emulations (VC 76 & VC 2A) conceived for Guitar Rig 4, like the other Studio Effects of the manufacturer. The good news is that the Guitar Rig 4 player is free. The bad news is that you can’t use the plugins unless you have the manufacturer’s guitar multi-effect. Available for Mac (Intel only) and PC in 32-bit and 64-bit versions, the bundle —including the host (GR4 or GR4 Player) plus the plugins— supports AU, VST and RTAS formats and includes a standalone version. As always, activation is done via the Service Center.

As their name already implies, IK Multimedia’s T-RackS Black 76 and T-RackS White 2A were conceived to work within T-Racks 3, but also as individual 32-bit or 64-bit plugins to be used directly in any DAW that supports AU, VST or RTAS. As usual, to activate them you’ll have to use the Authorization Manager.

As for Universal Audio’s UAD 1176LN and UAD LA2A plugins, they are only available for the DSP cards developed by the manufacturer, from the UAD-1 to the UAD-2 Quad. Our card uses OS 5.9.1. The plugins are Mac and PC compatible, they support 32-bit and 64-bit operation (only the card drivers actually work in 64 bits, the plugins still operate at 32 bits and require a bridge), and work with AU, VST and RTAS.

Did you ask for the 1176…?

All software manufacturers took their inspiration from the LN version of the FET 1176 compressor. LN (for Low Noise) means that the product includes a modification made by Mr Plunkett, engineer at Urei, who wanted to reduce the noise. You can recognize it by the famous black front instead of the traditional burst aluminum front with blue stripes around the VU-meter. These versions, referred to as C, D and E, are the most venerated. From a technical and audio standpoint, FET compressors contributed the principle of adjustable ratio, as well as much shorter attack and release times compared to competitors using Vari-Mu or optical designs.

IK Multimedia designed its plugin based on an E model. Neither Native Instruments nor Softube give any information on this matter. However, Softube’s experience developing the FET Compressor was certainly an advantage for Native Instruments. UA doesn’t give any information about the model either. Anyway, all three have a very similar sound character, just like signal processors that use only analog components (don’t forget that every unit sounds slightly different than the other, even if it’s the exact same version).

As for the features, except for the major innovation in the form of a stereo version (we’re talking about the 1176, not the Urei 1178 stereo version), UA stays faithful to the original. Thus, you get all the original features users like so much for their ease of use: a pair of big controls for input and output adjustment, two smaller Attack and Release knobs, four Ratio buttons (4:1, 8:1, 12:1, and 20:1) including an All-Buttons mode, and four VU-meter buttons (Off, +4, +8, and GR). However, you don’t have the possibility to set the Attack control to Off, which would switch the compressor to ratio 1:1, meaning you could process the signal without compression in order to get only the device’s sound character. All three manufacturers reproduced the (confusing, at least in the beginning) operation of the controls: the fastest attack time is not hard left (1) but hard right (7). The same applies to the release. Attack times range from 20 to 800 microseconds (yes, micro!), while release times range from 50 to 1,100 milliseconds.

Native Instruments and IK Multimedia added some modifications. IK Multimedia added four snapshots, taken from T-RackS’s architecture, plus L/R and M/S buttons allowing the user to choose between one of the two operating modes (well done!). Three additional buttons (L, R and =, where L and R become M and S in M/S mode) allow the user to process the two channels of a signal separately or together. IK Multimedia’s version also displays the setting values, but they don’t quite match reality, at least for Attack and Release, which is a pity. What’s the use of adding values if they have no meaning… The All-Buttons mode is accessible via a dedicated knob. Ratio 1:1 is available clicking the Off button under the Attack control. You also have Bypass and Reset buttons.

In Native Instruments’ version, a Ratio slider replaces all four original buttons but offers all usual values (4:1 to 20:1 plus All-Buttons mode). The 1:1 ratio replaces the bypass button under the Attack control. The VU-meter management also changed: you can display the input level, output level and gain reduction. The plugin comes with a preset menu accessible via the advanced settings. The side-chain input (great for techno/electro fans) comes with a slider that allows the user to adjust the amount of direct signal so that parallel compression is possible by adjusting the output level (which has no effect on compression itself). Unlike Softube’s FET Compressor, note that you get no continuous Ratio setting.

Now let’s take a closer look…

Conclusion

So, what should you choose, Native Instruments or IK Multimedia (UA doesn’t count because it was only used as a reference)? It’s a difficult question because both options provide advantages and good sound results. Will these plugins replace real 1176LN and LA-2A hardware processors? No. Does their performance match the original hardware versions? Yes, except for some applications (All-Button mode, transients management in given situations, settings and/or VU-meter calibration). Is it possible to do a good job with them? Yes. Software manufacturers benefit from the ever-increasing computer performance and offer more authentic emulations every time. Take for example the 24/7, which was considered a really good plugin when it was launched…

And these tools are affordable, which is not the case of the hardware gear they are based on. Each IK Multimedia module is sold for €89.99 and can be used in T-Racks. The Native Instruments bundle (including the three compressors) is sold for €199, while single plugins go for €99; and Guitar Rig Player 4, which is required for their use, is free. Moreover, both manufacturers offer fully-usable demo versions, which is an excellent way for you to expand on this comparison.

To read the full detailed article see: Vintage Compressors Review

July 28, 2011

Mysteries of Dynamics Processing Revealed

Filed under: Compressors, Processors, Recording reviews — Tags: , , , , , — audiofanzine @ 7:34 pm

A dynamic processor is something that outputs a signal, where the level of the outgoing signal is based on the level of the incoming signal. In other words, a loud signal coming in will come out differently than a quiet signal coming in.

Basic types of Dynamic Processors

Compressors: The most common – the louder the signal is coming in, the less level it provides going out. In a compressor, a target level is set – called the “threshold” – and any signal coming in that exceeds that level will be reduced. The higher the level is above that threshold, the more reduction will occur. More on this later.

 

Limiters: Limiters are like super compressors. The idea is to ensure that the level does not exceed the threshold. Because this amount of compression is extreme, a limiter relies on certain functions and design that regular compressors do not have.

Expanders: The quieter the signal is coming in, the less level it provides going out. In other words – it makes quiet signals even quieter. Much like a compressor, the threshold is set at a certain level. Any signal that does NOT exceed that threshold is reduced, and the quieter the signal, the more reduction is done.

 

Gates: Gates are like super expanders. Anything that does not exceed the threshold is reduced to inaudible. Again, because gates are extreme, they often require a slightly different design than a regular expander.

 

Now – I’ll focus primarily on Compression, because that’s going to be the most commonly used dynamic processor.

Compression

Every signal you hear is compressed??? Yes, every signal you hear is compressed.

Bare with me. Imagine you have a rapper in front of a microphone. The rapper raps, you record. You play it back. You haven’t used any processing – you’re just playing back the raw vocals.  You are listening to a signal that has gone through at bare minimum 3 stages of compression – and more likely than not – closer to 6.

  • The microphone capsule gains tension as the rappers voice pushes it – in other words – it pushes back. The more the rapper’s voice pushes in – the harder the capsule diaphragm pushes back. In other words, the louder the signal is hitting the capsule, the more reduction the capsule does to the signal. That’s compression! (It’s mild compression, but it’s still compression).
  • Along the way through the microphone, you may hit a tube. Tubes have a non-linear response to voltage – the response is quite curved, and also changes the frequency balance of the signal. This is called saturation – which will tend to “round out” a signal, by reducing the loudest peaks. Compression! And before leaving the microphone, the signal may hit a transformer as well, which will saturate in a similar way… more compression.
  • The preamp is going to have multiple stages of saturation – and often times, the more gain you give something – the deeper that saturation curve goes. In other words, the more you drive the signal at the preamp, the more compression the signal experiences.
  • Then the sound has to actually come out of the speaker cones. Well, those speaker cones are going to build up tension when pushed further. See where this is going? This is called “cone compression”.

Ok – so this is a bit of a simplification – but there’s a point here. The point is that “compression” is always part of the signal. Some mics have less of it, some have more – same with speakers, tubes, transformers, etc. And they all do it in different ways. With tubes, people will talk about their saturation curves and %THD (total harmonic distortion – or frequency alterations). With mics, people will refer to how it “grabs” a sound – or more specifically – the sound’s shape.

Now let’s take a closer look…

Maximum Punch

There is a thin line between a transient sound, and a sustained sound. A sound that holds for any noticeable amount of time is sustaining. A sound that moves by too quickly to register as it’s own moment is transient. But transients can vary in length. A transient can be half a millisecond or it could also be ten milliseconds; they won’t sound the same. A big factor in punch is how long that transient exists. A quick transient sounds “spikey” – but a long transient sounds “punchy.” You want to find the point that makes the transient exist as long as possible before “flattening out” or becoming a sustained sound. Only your ear can tell you where that point is.

 

Good samples are already shaped to have that kind of impact – and any additional compression may actually soften that. Of course, punch has a lot to do with frequency as well – but that’s for another article.

 

Now what about the release? The release is super elusive. It determines how long it takes for the compressor to let go. If the release is too short for the signal you are going to get a disjointed sounding shape which usually results in distortion. If it’s too long, your signal never really returns to its natural shape, and you generally lose tone (or you just get permanent drive on the compressor’s output, giving the whole signal a new bit of tone). So the idea is to find a point that emphasizes the sustain (which is where most of the signals tone lives) properly.

 

Lastly, when the attack and release are set in a way that seem to argue – the compression can become very audible. You either hear the decent or the ascent of the signal level. This is called pumping. It’s generally annoying, but can sometimes be used an effect. If audibly desired, consider the rhythm of the release time, and ask yourself if it’s groove is complimenting the song.

———————————-

So, rather than thinking of a compressor as something that effects the “level” of a signal. Think of a compressor as something that effects shape. Why? Because level can be controlled with the volume fader more accurately and transparently. A fader doesn’t really control shape, unless you are being extremely meticulous. Conversely, compression will always effect the shape of the sound it is working on.

Once you start hearing shape, you will understand compression.

To read the full detailed article see:  Mysteries of Dynamics Processing Revealed

February 4, 2011

Dynamics Processing Meets Rock Guitar: How to Compress a Guitar or Bass

Dynamics processing with studio-oriented processors? Been there, done that. But have you re-visited it lately in a guitar context? Dynamics control for vocals or program material is very different compared to guitar. Much of this is because there are many ways to use dynamics processing for guitar (or bass). So, let’s take a look at the different ways to apply dynamics, with examples of suggested settings.

For an introduction to compression, check out the article “Compressors Demystified.” If you’re already up to speed, let’s give a few basics on how to set up studio processors with guitar (however, note that these same basic techniques work with plug-in software compressors as well as hardware).

The Interface Space

“Stomp box” dynamics processors, while designed specifically for guitar, are more limited than rack-mount studio hardware – but the latter have issue levels with guitar. Interfacing involves one of four approaches:

Use the instrument input. If the processor has an “instrument” input, you’re golden. Plug the guitar directly into the processor, then run it into the mixer, amp modeler, guitar amp (assuming you can adjust the output level to avoid total overload), or whatever. Look for an instrument input impedance above 100kilohms, and preferably above 220kilohms, to avoid dulling high frequencies and reducing level. But too high an impedance (in the 5-10Megohm range) reaches a point of diminishing returns, because now the input may be too sensitive and prone to noise pickup. A 1Megohm impedance is a good compromise setting.

Use a preamp or suitable direct box. Adding a preamp or direct box (assuming it has an appropriately high input impedance) before the processor will preserve the guitar signal’s fidelity and allow for best level matching. If you’re driving a guitar amp, you may be able to use the dynamics processor’s output control to add some extra overdrive, but don’t go overboard (or do, if you like really nasty sounds!).

Insert into your guitar amp’s effects loop. If you want to record with your guitar amp but are using a line-level processor, patch it into the guitar amp’s effects loop. The loop should be able to provide line levels for the send (goes into the processor’s input) and return (comes from the processor’s output).

If you’re using a hardware mixer, insert the dynamics processor into your mixer’s channel inserts. This will also match levels properly, although you’ll still have to figure out how to interface the guitar with the mixer. The choices are the same as above: If the mixer has an instrument input, great. If not, use a preamp, direct box, etc. between the guitar and mixer.

Now let’s take a closer look how to really do it…

Double Your Pleasure

Patching two compressors in series, with both set for small amounts of compression, can give a significant amount of compression but sound less obvious than using a single compressor to give the same amount of compression. The first stage essentially “pre-conditions” the signal so that the second compressor doesn’t have to work so hard.

 

If you have a stereo compressor that can be set to dual mono operation, you can patch the two individual compression channels in series. With plug-ins, you can just insert two in series in a track. The drawback is that unlike standard compression, where you have to adjust only one set of controls, an ˆ la carte approach requires adjusting both sets of compressor controls. While this might seem like a disadvantage, most of the time you’ll set them to similar settings anyway.

Window Shopping

To get an idea of what’s out there in compressor-land, visit a few retailers and manufacturers and you’ll see the choices are huge, ranging from under a hundred dollars to thousands (and thousands!) of dollars. But realistically, for the type of application we’re describing here, you don’t need anything too fancy – it’s not like you’re using the compressor to re-master vintage recordings for audiophile releases. Besides, these days technology is at a level where even fairly inexpensive devices can deliver excellent results.

 

In any event, all the above tips are just guidelines. Experiment with your dynamics processor, and you may find yet another way to exploit these perhaps unglamorous, but extremely useful, devices.

To read the full detailed article see:  How to Compress a Guitar or Bass

February 2, 2011

Mastering: Curve Analysis and Acquision Software

Bob Ludwig, Doug Sax, Bernie Grundman – they’re masters of mastering. They produce hit after hit, with nothing at their disposal other than…well, experience, talent, great ears, the right gear, and superb acoustics.

So maybe you’re missing one or more of those elements, and wish that what came out of your studio sounded as good as what comes out of theirs. So, why not just analyze the spectral response curves of well-mastered recordings, and apply those responses to you own tunes?

Why not, indeed – but can you really steal someone’s distinctive spectral balance and get that magic sound?

The answer is no…and yes. No, because it’s highly unlikely that EQ decisions made for one piece of music are going to work with another. So even if you do steal the response, it’s not necessarily going to have the same effect. But the other answer is yes, because curve-stealing processors can really help you understand the way songs are mixed and mastered, and point the way toward improving the quality of your own tunes.

As to the tools that do this sort of thing, we’ll look at Steinberg’s FreeFilter (which was discontinued, but still appears in stores sometimes), Voxengo CurveEQ, and Har-Bal Harmonic Balancer. They’re very similar, yet also, very different.

How They Work

FreeFilter and Voxengo split the spectrum into multiple frequency bands in order to analyze a signal. These create a spectral response, as from a spectrum analyzer, while a song plays back. During playback, the program builds a curve that shows the average amount of energy at various frequencies. You can apply this analysis (reference) curve to a target file so that the target will have the same spectral response as the analyzed file, as well as edit and save the reference file.

Har-Bal isn’t curve-stealing software per se. While optionally observing the response of a reference signal, you can open another file, and see its curve superimposed upon the reference. You can edit the opened file’s curve so it matches the reference signal more closely, but this is a manual, not automatic, process.

Fig. 1: The black line is the spectral response for Madonna’s Ray of Light; the red line represents a Fatboy Slim mix. Fatboy’s has a lot more treble, while Ray of Light has a serious low-end peak.

The manual vs. automatic aspect is in some ways a workflow issue. FreeFilter and Voxengo start by creating the reference curve, but give you the tools to adjust this manually because you’ll probably want to make some changes. Har-Bal takes the reverse route: You start out manually, and if you want to, use the tools to create something that resembles the visual reference curve, which was generated automatically when you opened the file. Also remember that curve-stealing is only a part of these programs’ talents; they’re really sophisticated EQs.

So what do some typical curves look like? Check out Fig. 1. The black line is the spectral response for Madonna’s “Ray of Light,” while the red line represents a Fatboy Slim mix. Past about 1 kHz, Fatboy’s curve shows enough high frequency energy to shatter glass. “Ray of Light” has a higher response below about 400 Hz, due mostly to a prominent kick. It has a more thud-heavy, disco kind of vibe, whereas Fatboy Slim leans more toward a techno style of mastering. Apply these curves to your own music, and they’ll take on the characteristics of the reference tunes – but the results may not be what you expect, as we’ll see.

Now let’s take a look at the individual software…

So What Does Work?

Using your ears to compare your work to a well-mastered recording is a tried-and-true technique, but it shortens the learning process when you can actually compare curves visually and see what frequencies exhibit the greatest differences.

I’ve found a few reference comparison curves for Har-Bal that work well for certain types of music: Fatboy Slim for when dance mixes are too dull, “Ray of Light” for a house music-type low-end boost, Cirque de Soleil’s “Alegria” for rock music, and Gloria Estefan’s “Mi Tierra” for acoustic projects. On very rare occasions I use their curves, but when I do, they’re more like “presets” because they end up getting tweaked a lot. Automatic curve-stealing just doesn’t do it for me, but “save me 10 minutes by putting me in the ballpark” does.

But my main use for curve-analyzing software is for stealing from myself. After mastering a music project for a soundtrack, one tune sounded a little better than the others – everything fell together just right. So, as an experiment, I subtly applied its response to some of the other tunes. The entire collection ended up sounding more consistent, but the differences between tunes remained intact – just as I’d hoped.

Another good use was when German musician Dr. Walker remixed one of my tunes for a compilation CD, but used a loop for which he couldn’t get legal clearance. Rather than give up, I created a similar loop that wasn’t a copy, but had a similar “vibe.” Yet it didn’t really do the job – until I applied the illegal loop’s response curve to my copy. Bingo! The timbral match was actually more important than the particular notes I played in terms of making the loop work with the rest of the tune.

To read the full detailed article please see:  Curves of Steal

This does produce a weird paradox, though: I used a piece of curve-stealing software to avoid stealing a piece of copyrighted material. I guess it’s all part of the living in the 21st century.

December 30, 2010

Compressors: How They Really Work

It’s one of the most used, and most misunderstood, signal processors. While people use it to make a recording “punchier,” it often ends up dulling the sound instead because the controls aren’t set optimally. And it was supposed to go away when the digital age, with its wide dynamic range, appeared.

Yet the compressor is more popular than ever, with more variations on the basic concept than ever before. Let’s look at what’s available, pros and cons of the different types, and applications.

Introduction

Compression was originally invented to shoehorn the dynamics of live music (which can exceed 100 dB) into the restricted dynamic range of radio and TV broadcasts (around 40-50 dB), vinyl (50-60 dB), and tape (40dB to 105 dB, depending on type, speed, and noise reduction used). As shown in Fig. 1, this process lowers only the peaks of signals while leaving lower levels unchanged, then boosts the overall level to bring the signal peaks back up to maximum. (Bringing up the level also brings up any noise as well, but you can’t have everything.)

Fig. 1: The first, black section shows the original audio. The middle, green section shows the same audio after compression; the third, blue section shows the same audio after compression and turning up the output control. Note how softer parts ot the first section have much higher levels in the third section, yet the peak values are the same.

Even though digital media such as the CD have a decent dynamic range, people are accustomed to compressed sound. Compression has been standard practice to help soft signals overcome the ambient noise in typical listening environments; furthermore, analog tape has an inherent, natural compression that engineers have used (consciously or not) for over half a century.

There are other reasons for compression. With digital encoding, higher levels have less distortion than lower levels—the opposite of analog technology. So, when recording into digital systems (tape or hard disk), compression can shift most of the signal to a higher overall average level to maximize resolution.

Compression can create greater apparent loudness (commercials on TV sound so much louder than the programs because they are compressed without mercy). Furthermore, given a choice between two roughly equivalent signal sources, people will often prefer the louder one. And of course, compression can smooth out a sound—from increasing piano sustain to compensating for a singer’s poor mic technique.

Now let’s look at some compressor basics…

Compressor Types

Compressors are available in hardware (usually a rack mount design or for guitarists, a “stomp box”) and as software plug-ins for existing digital audio-based programs. Following is a description of various compressor types.

  • “Old faithful.” Whether rack-mount or software-based, typical features include two channels with gain reduction amount meters that show how much your signal is being compressed, and most of the controls mentioned above.
  • Multiband compressors. These divide the audio spectrum into multiple bands, with each one compressed individually. This allows for a less “effected” sound (for example, low frequencies don’t end up compressing high frequencies), and some models let you compress only the frequency ranges that need to be compressed.
  • Vintage and specialty compressors. Some swear that only the compressor in an SSL console will do the job. Others find the ultimate squeeze to be a big bucks tube compressor. And some guitarists can’t live without their vintage Dan Armstrong Orange Squeezer, considered by many to be the finest guitar sustainer ever made. Fact is, all compressors have a distinctive sound, and what might work for one sound source might not work for another. If you don’t have that cool, tube-based compressor from the 50s of which engineers are enamored, don’t lose too much sleep over it: Many software plug-ins emulate vintage gear with an astonishing degree of accuracy.

Whatever kind of audio work you do, there’s a compressor somewhere in your future. Just don’t overcompress—in fact, avoid using compression as a cop out for bad mic technique or dead strings on a guitar. I wouldn’t go as far as those who diss all kinds of compression, but it is an effect that needs to be used subtly to do its best.

To read the full article see:  Compressors Demystified

August 12, 2009

EQ and Compression Techniques Pt.2: Drums

Despite the preponderance of exceptional drum samples and loops on the market, for certain genres of music (notably country and rock) there is no substitute for a great session drummer playing on a well-recorded and mixed drum kit. One thing that samples and loops can’t provide is the great rhythmic instincts an accomplished live player draws upon when responding to a specific song. However, getting a great player (while certainly a significant element) is not the entire story. The appropriate treatment of the drums in a mix with EQ and compression can make the difference between a lifeless, vague sound and an exciting, textured and genuinely rhythmic drum track.

Even though the drummer plays the entire kit as a single instrument, the miking of individual drums and cymbals can make for a very complicated mix scenario. The reason I reference country and rock music specifically has to do with the fact that in these genres the sounds of the individual drums and cymbals are not only singled out by individual microphones placed on each of them but also their sounds are exaggerated to create an even more dramatic effect. Consider, for example, the tom fills in Phil Collins’ “In The Air Tonight.” By contrast, jazz drums are often treated as a more cohesive, unified sound and it’s not unusual to use a simple pair of overhead mics to capture the sound of the entire jazz drum kit.

In this article, I’m going to go drum by drum providing EQ and compression settings that will, hopefully, provide you with a jumping off point to getting great drum sounds in your mix. Because of its all-in-one mixing board channel approach, I’ll be using Metric Halo’s Channel Strip plug-in with its EQ, compression and noise-gate to illustrate my comments about various EQ and compression settings.

Now lets take a closer look drum by drum…

Conclusion

While I’ve been painfully specific about EQ, compression and gate settings, it’s important to remember that every mix situation is different. Use all of these settings as a jumping off point and then use your ears to tweak the sounds until you’re happy. Good luck!

To read the full detailed article see:  EQ and Compression Techniques for Drums

June 30, 2009

Charter Oak – SCL1 Compressor Limiter

Charter Oak gives us an exclusive presentation of their new discrete Compressor Limiter, the SCL1.

To see more exclusive video demos visit Audiofanzine Videos.

Older Posts »

Create a free website or blog at WordPress.com.