Mastering EQ

EQ the Midrange

Listen and try to identify any problems that you hear. Start with the midrange (vocals, guitar, midrange keyboard, etc.) as this will typically represent the heart and soul of the song. Does it sound too "muddy"? Too nasal? Too harsh? Compare it to another mix, perhaps a commercial CD. Try to describe to yourself what the difference is between the two mixes around the midrange.

Too muddy?

Try cutting between 100 to 300Hz

Too nasal sounding?

Try cutting between 250 to 1000 Hz.

Too harsh sounding?

This can be caused by frequencies in the range of 1000 to 3000 Hz. Try cutting this range a few dB.

Hopefully, using a band or two in these regions will give you a better sounding midrange.

You'll get the most natural sound using wide bands (Q less than 1.0). If you find yourself using too narrow a notch filter, or too much gain, you may be trying to fix something that EQ on a stereo mix can't fix. Go back to the individual tracks and try to isolate the problem that way.

Note also that the wider the band, in general the less gain you need to apply. In addition, your ears quickly get used to EQ changes. You may find yourself boosting more than necessary to hear the difference.

EQ the Bass

A reasonable use of EQ in the low end is to shelf filter below 30-40Hz. Purists might find this alarming, as yes, we can hear down to 20Hz and some musical information can be lost. Typically what people consider "bass" though is in the 50-100Hz region, and the audio in the 20-40 Hz range can usually be rolled off. The benefit is that you can remove some low frequency rumble and noise that could otherwise overload your levels.

Keep in mind that for bass, or any EQ change for that matter, every action has an opposite reaction. If you increase one frequency, you can mask another frequency. The flipside of this is that cutting one frequency can be perceived as a boost to another frequency. Each change that you make can affect the perception of the overall tonal balance of a whole.

Bass guitars and kick drums can span a wide frequency range. Where the "oomph" of the kick drum can be centered around 100Hz, the attack is usually found in the 1000-3000 Hz region.

Sometimes you can get a sharper sounding "bass" sound by focusing on the higher frequency attack, as opposed to the 100Hz region which can cause "mud". On the other hand, if you want to add that hip-hop style "ring" to the bass, try a peak at 50-60Hz.

EQ the Highs

Finally, take a listen to the higher end frequencies in your mix.

Summary of General EQ Tips

EQ Chart

These settings are just a starting point, but are pretty useful as long as you trust your ears more than your eyes…

Mastering EQ Chart

These frequencies are very generalised, but useful in altering the tonal balance of a mixed track.

© Matt Bellingham 2003 – 2006

MASTER CLASS

A different way of looking at compression.

This article was originally published in Studio Sound magazine.

IN AN IDEAL WORLD each stage of the recording / transmission / reproduction chain would have an equally wide dynamic range capability. In reality the dynamic range of each subsequent stage is generally less than that of the preceding stage. The average analogue console has a usable dynamic range of about 110 dB. Plain 16 bit digital audio has typically less than 96 dB. FM radio has around 55 dB and the worst case listening environment, the car, can fall to 20 dB or less! The dynamic range capability of the human ear is about 120 dB although there may actually be a shifting range of around 60 dB if other factors are considered. In this article I will look at a way of reducing dynamic range to suit FM radio and particularly how to implement this during recording and post-production. The system does have other uses which will also be examined.

The difficulty in broadcasting is that there is a strict limit on the transmittable dynamic range. This can be broadly defined as the difference between the transmission systems noise floor and the point of maximum modulation. In FM radio it is around 55 dB. Some classical music can exceed this figure and so it becomes desirable to reduce its dynamic range. The actual usable dynamic range is ultimately dictated by the listener's environment. Control of the station's dynamic profile to suit this environment is really the domain of station processing. Later I will briefly discuss the possibility of using this compression system in that application.

BEFORE COMPRESSION WAS AVAILABLE engineers had to tailor dynamic range by hand. This involved following a score (or working from memory) and making level changes in anticipation of the peaks and troughs in the music. These changes would be made as slowly as possible and resulted in very transparent dynamic control. I am sure many readers will remember doing this in the old 'AM days'. Interestingly, this method mimics approximately the apparent dynamic response of the ear[1]. For subtlety, you still can't beat a sensitive engineer with a good set of ears.

Figure 1 - Standard compressionAs technology advanced, the range and quality of electronic compression devices improved drastically. Even so, one of the main objections to the normal compressor configuration (fig. 1, at left) is that it reduces the impact of loud passages by 'holding them back'. This is particularly noticeable when these passages contain transient material such as percussion. It is the departure from a 1:1 gain law at high levels and the effect this has on the impact of the music that is most noticeable. This, combined with the apparent lifting of low-level passages caused by slow release times and gain make-up after the compressor, has probably generated more letters to the editor than any other broadcast topic!

Some manufacturers have tried to address these problems with multi-stage or multi-band techniques and programme-dependent attack and release times. On the whole these are an improvement, especially where there is time to fine tune the device to suit the audio being processed. The BBC has even developed a system called 'DRACULA' that uses a three second look ahead to try and better place level alterations in the correct dynamic context[2]. However, there will always be a section of the public and industry that finds the action of these devices disturbing. I think I can safely say that the listeners most likely to notice the effects of dynamic controllers tend to be the ones that listen to the types of music that defy transparent use of those controllers! The following compression system is subtle enough to go largely unnoticed yet still provide 'real' amounts of control.

SOME YEARS AGO IN Studio Sound magazine [3] a colleague found a brief description of the method presented here. It has turned out to be such a useful tool that I am presenting it again, this time in more detail and with some refinements. In addition I will examine other uses and implementations of the system. For those wanting a fuller understanding there is a mathematical treatment.

Figure 2 - Side Chain compressionThis system relies on a secondary compressed signal path running in parallel with the main signal path (see fig 2, at right). This basic idea is not new. It has been around since at least the early sixties and dozens of companies have used it as the basis of a wide range of products. The most well known user of the technique is Dolby Labs, with a string of patents as long as your arm. A colleague dubbed our implementation 'Side Chain Compression' or simply 'Side Chain'. These terms should not be confused with the internal control side chain of the compressor itself. This name has stuck and I will use it here, although in retrospect 'dynamic scaling' might have been better.

Setting up the system is fairly easy. In my case I use a secondary stereo buss to feed the compressor (fig 3, at left). Any audio to be processed is sent to the main and secondary busses. A stereo channel is used to return the compressor's output to the desk which in turn is sent to the main buss only. The return must be in phase with the original signal.

Calibration is also straightforward and enables the user to achieve consistent results at different locations. 1 kHz tone at reference level ( 0dBr ) is fed to the Side Chain buss only. This would normally be 0 VU or PPM 4. The compressor is set for 16 dB of gain reduction at a 2:1 ratio. Fast attack and slow release times should be used and in my case these are 2.5 mS and 800 mS respectively. Release times as short as 25 mS can be used depending on the type of music and the effect required. Experiments with digital compressors within ProTools have shown that times shorter than these don't really work because they follow the waveform too closely.

Figure 4 - Fader calibrationThe level at the compressor's output is made up to 0, in this case requiring 16 dB of gain. The compressor's return channel is trimmed so that a setting of 0 on the fader gives a reading of 0 (dBr) on the main meters. In order to get a consistent indication of the amount of Side Chain the compressor's return channel can be marked with a chinagraph (grease) pencil as in figure 4 (the fader, at left). To get these points, put tone back on both busses and add enough Side Chain level to increase the level on the main buss by 0.25, 0.5, 0.75, 1, 2, and 3 dB.

Figure 5 - 1 dBSCFigure 5 (on the right) shows the effects of '1 dB' of Side Chain compression (dBsc). Through the middle is a 1:1 gain law representing the main signal path. Below is the output of the compressor at 1 dBsc (-18 dB set-point). Above is the resultant slope - a ratio of 1.14:1. As can be seen there is approximately 5 dB of lift at the -32 dBr threshold. A linear transfer curve (1:1 offset by 5dB) is maintained below this point. Figure 6 shows the actual plots of the system I use. The main difference is the curve at the threshold due to the compressor's 'soft knee'. Because the system has 1 dB of 'gain' this can be corrected at the master faders. This shifts the slope down by 1 dB giving +4 dB at -32 dBr. In practice it is only necessary to reduce the masters when the amount of Side Chain exceeds 1 dBsc. The calibrated scale outlined above will show exactly how much this reduction should be.

The original 1977 description of this method called for 20 dB gain reduction. In our case we were using Neve 2254s which have only 16 dB of metered GR available and this became our 'standard'. See table 1 for a comparison of the two. It is possible to implement the system on some digital audio workstations.

Figure 7 shows Side Chain at 1 dBsc compared with 2:1 compression. As can be seen, the low level lift at the Side Chain's threshold of -32 dBr is the same in both cases. Both methods achieve 4 dB of gain reduction. In the case of standard compression this is achieved by scaling the top 8 dB at 2:1. The Side Chain method spreads this gain change over 32 dB, giving a ratio of 1.14:1. This is achieved by adding a highly compressed version of the signal at low level with itself. At high input levels the effect of the compressed path is small - nil if you reduce the master faders as detailed earlier. As the level drops, the additive effect of the Side Chain path increases and this is what gives such a gentle slope.

There are two main reasons why Side Chain compression sounds better. Firstly the gain law is gentler when it departs from 1:1. In fact, the threshold is so low that the system is in 'gain reduction' most of the time. Secondly, the method of implementation takes the compressor out of the main signal path. Standard compression derives its control directly from the signal and this control is applied via a feedback loop to the signal itself. With Side Chain, the main signal still 'controls' the amount of gain change but the gain law is derived indirectly by signal addition. Purists should still expect some smearing of transients due to non-linear group-delay in the compressor and return channel but the effect of this is insignificant when compared with a compressor in the main signal path.

Like a standard compressor, the correct level must be fed into the system for it to be effective. The audio should be peaking to the correct level on the main meters before the Side Chain is added. Be warned that over-driving the system is likely to push the compressor beyond its gain reduction range and could result in distortion. Under-driving it increases the effect of the Side Chain by effectively raising the threshold. This is the opposite of the effect from standard compression under the same circumstances. As a rule, if the maximum peaks hit PPM 6 (0 VU) the rest of the dynamic should take care of itself.

Provided these steps have been followed the amount of Side Chain to be added is up to the user. For classical music I have found the aesthetic maximum to be about 1 dBsc but it also depends a lot on the instrumentation. For example, solo harpsichord never needs Side Chain because its highly transient nature makes it sound loud even at low levels. A full orchestra performing dynamic repertoire can accommodate 1 dBsc or more. If there are extended low level passages I prefer to ride the master faders rather than adding more Side Chain since noise can become obtrusive at higher settings.

The great thing about Side Chain compression is that you can alter the amount you add in real time and those alterations can be very subtle. This is useful when recording or broadcasting public concerts, as you can remove the Side Chain in between movements so that audience noise is not increased. On the other hand, it can be used to create a 'cushion' of atmosphere which sits nicely under announcements. The other advantage of using it is that low level passages can be lifted without the risk that unexpectedly loud passages will get out of hand. In the case of live broadcasts I don't use it on the announcer - standard compression set for a medium attack and a slow release is more appropriate if it's needed at all.

As a general rule I also remove the Side Chain at the ends of pieces unless the audience sounds a bit thin. In that case I use it to increase the density of the applause. Because I use a secondary buss to feed the Side Chain it is possible to pick exactly what is processed. I don't send reverb or ambience mics to it but that's only a personal preference. At the suggested setting the audible side-effects are virtually nil. Like any form of compression though, it increases the density of the sound and this is particularly noticeable in the reverb. It can also appear to change the balance between a soloist and accompanying ensemble. This subtle lifting of low level information is very difficult to detect in isolation so I would suggest that you include its use in your session notes. Ultimately, because the quiet material is lifted, the average level increases and the ear perceives that the sound is louder and somehow 'better'[4].

It is possible to use other compression ratios and different amounts of gain reduction. Even devices that have soft knees or varying ratios can be used. The only proviso is that attack and release times should be fast enough to closely track the programme's dynamics. See the side bar for a full mathematical treatment.

WHILE I USE Side Chain compression primarily on classical music there are some other areas where it could be useful. The system could be used as the main on-air processor for FM radio with a couple of caveats. Because it reduces the dynamic where the peak level has already been correctly set it is not suitable as a gain-riding processor. It is best used where the levels are already being properly set such as on a dual operation type station with a separate engineer and announcer. Because the effect is so subtle the result is very listenable, even when set as high as 3 dBsc. At this level care is needed setting the release time. If a fixed amount of Side Chain were used all the time the station could offer a 'decoder' of some sort to reverse the compression for critical listeners. As with any over-all station processing, the level of vocal (spoken) material may have to be reduced slightly to maintain the correct on-air balance between it and music.

This mode of compression works well on non-classical music too - both on whole mixes and individual tracks. I have found that 2-3 dB of Side Chain compression applied to a mix destined for AM radio creates a denser sounding track without affecting the transients too much. The result of this 'pre-processing'is a track which is less likely to be mangled by the station's own on-air processing. 3 dBsc or more can be used to make really dense sounding tracks and yet the ratio is still only 1.36:1. It may be necessary to slow the release time a little depending on the desired effect.

Standard compression can be used quite creatively when tracking[5]. It is also relatively transparent at this stage as each source is processed individually. Side Chain can be used where very transparent control is needed. Transients are retained while the dynamic can be precisely controlled to make it sit in the mix. While it works well with most instruments it doesn't work so well with singers. If you want to try it for vocal work, it would be advisable to use a peak limiter prior to tape to catch any big peaks. The vocal application in which it really excels is the recording of speech. I have used it for documentaries and the like to reduce the progressive reduction in level that can happen during long reads. This gives a more natural and open sound during lengthy passages of speech.

I have also found it a useful mastering tool - particularly for classical releases destined for cassette. The whole dynamic can be scaled to fit comfortably on cassette without audio being buried in tape noise a lot of the time. I also subtly peak limit to avoid saturation. I then let the cassette duplicator know where the biggest peaks are, thus guaranteeing a hot print on that medium. The CD version is printed without it of course. I have also occasionally used it when mastering pop/rock albums where the effect seems appropriate. Ultimately, it depends on what you want to achieve and what sounds best[6].

In pure mastering applications it is best to remove the Side Chain as the track fades or dies away. This is because it causes a slight lift in the noise floor as the audio disappears.

As with any compression system there are side effects which can be useful in themselves. In one case I used the system to repair the balance of a classical concert with solo vocalist. It was the kind of recording that post-production engineers have nightmares about. Individually the orchestral balance was fine as was the sound of the vocalist. Unfortunately the vocalist was far too loud and the recording was very dry. I added 3 dB of Side Chain compression with a fast release time. This had the effect of lifting the orchestra in between vocal passages. Two reverbs were used. One send was EQed to have mostly voice and the other mostly orchestra. The orchestral reverb was set in such a way that it carried the sound through the start of vocal lines - hiding the fact that the overall level had ducked. This relatively severe treatment was acceptable for a one-off broadcast but would have been too much for a CD release. The untouched audio was broadcast with pictures elsewhere and listeners to radio were surprised at how much better 'our' sound was.

THERE ARE A COUPLE of other things you might like to try. Putting a bass cut in the Side Chain return reduces the lifting of low frequency noise. In some cases I have EQed other areas for various effects. Boosting at 100 Hz and 10 kHz (both peaking) can give you a subtle contouring effect at low levels. You could also use a multi-band compressor like MDT[*] in ProTools to create subtle dynamic equalisation effects. More severe EQ can be used to lift instruments in a mix.

One commercial production operator I knew, used to EQ the feed to the compressor. He would boost the speech intelligibility frequencies only and this gave immense HF density without resorting to lots of EQ in the main signal path.

The other thing that you can try is to send from the Side Chain channel to your reverb. The send-level can even be set slightly higher than those of your audio source/s. This creates a cushion of reverb that seems to 'stick' better - especially with cheaper reverb units. It also stops recordings 'drying up' as the level decreases. I also insert a compressor in the send to the reverb to flatten quick peaks. This reduces excessive reverb hang-over during repetitious staccato passages

It would also be possible to process the sum and difference signals rather than the discrete left and right signals. The effects of fixed processing of sum and difference signals have been covered in Studio Sound several times[7, 8]. The idea with processing the sum and difference signals would be to enhance or alter the image width. It's probably more useful as a special effect than for serious work.

This system of compression provides an alternative to regular compression where subtlety is required and is a useful tool to have in one's 'box of tricks'. I would welcome feedback from anyone who tries this system themselves or has any suggestions.

Richard Hulse is a senior recording engineer with Radio New Zealand Limited. He can be contacted at e-mail rhulse@radionz.co.nz or Radio New Zealand, P.O. Box 123, Wellington, New Zealand.

This entire document © 1997 Richard Hulse. Reproduced here with the author's permission.

Common Frequencies For Equalization

FREQUENCY:

USES:

50Hz

100Hz

200Hz

400Hz

800Hz

1.5KHz

3KHz

5KHz

7KHz

  10KHz

15KHz

What is compression? 

Compression is the process of lessening the dynamic range between the loudest and quietest parts of an audio signal. This is done by boosting the quieter signals and attenuating the louder signals. The controls you are given to set up a compressor are usually:

Compression Types

Compressors come in various different flavors. These are used by engineers for different tasks and some sound far better in certain situations than others.

VCA Compression

Voltage Controlled Amplifier compressors use an integrated circuit to give very precise control. They are less colored and suffer from very few side effects like distortion, which make them ideal for lots of different tasks. The dBx 160 is a VCA compressor.

Opto Compression

Opto, meaning optical, describes the light sensitive circuits that control the compression amount in opto compressors. They often react more slowly than other compressor types, but this can be desirable. The famed Teletronix LA2A is an optical compressor that many producers swear by for vocals and mix bus compression. The LA2A is also a ‘leveling amplifier’ — which means it is working nearly all the time, not just when a threshold is reached.

FET Compression

Field Effect Compressors use transistors to emulate a valve sound with more reliability, but with a higher signal to noise ratio. They are popular for vocals and great for drum compression. The Urei 1176 is a FET compressor.

Valve Compression

Valve compressors work in one of the three ways described above, but use valves in the amplifier circuit to get that ‘creamy’ sound. The LA2A, which is an opto compressor, uses valves.

How Set Up a Compressor

1. Whether you’re using a hardware compressor or a plug-in, setting up works the same way. Insert the compressor on the channel you want to compress.

2. Adjust the threshold until the peaks in the signal are pushing over the threshold and triggering the compressor. Unless, of course, you really want to clamp something—like a live bass maybe—in this case it can work to make it push over the threshold all the time.

3. Set the Ratio to suit the material. Bass guitars sound good at 4:1, drums at 2:1, vocals also at 2:1 and electric guitars anywhere from 2:1 to 6:1.

4. The Ratio and Threshold work together. Adjust them together and see how they affect the output.

5. The attack and release controls shape how the compressor reacts. A fast attack would be useful for a rapper or anything that has sudden peaks early in the signal. Slower attack times suit mastering uses and buss compression.

6. The release control can really affect the sound of the compressor. Short release times cause the compressor to sound like it’s working hard, but long release times sound more natural.

7. Use the make-up gain and output control to sit the signal back into the mix without adding any unnecessary noise.

8. Setting the hard/soft knee would depend on the material. Hard knee works well for drums, bass and percussive stuff. Soft knee is more transparent and better for vocals and some guitar parts.

9. Look-ahead. Plug-in compressors often have this feature. It uses a slight time delay on the whole song to give the compressor a sneak preview of what’s coming. This allows it to catch all the peaks in the smoothest possible way. It can sometimes cause the compressor to lose its ‘character’ so don’t use it by default—only if necessary.

For every rule about setting up compressors, there’s someone who has broken the rules and made a great sounding record, so experiment.

A final word of warning—compressing on the way to your recording format, be it tape or hard disk—can’t be undone. Use compression sparingly whilst recording. Save it for the mix until you’ve got enough experience to know you’re not overdoing it.

http://audio.tutsplus.com/author/sean-vincent/

	Inside EQ - The Keys To A Great Mix by Jake Hartsfield Jake Hartsfield is a songwriter, producer, touring sound engineer and a new addition to the TuneCore Marketing Team. With today’s technology – GarageBand, for example – anyone can record a track straight into their computer using their computer mic, slap on a few automatic effects, and instantly have a finished recording of a song.  If you try to transition from Garage Band to Logic or Pro Tools, it can be a little overwhelming trying to understand all of the fundamental rules of mixing that are done for you in Garage Band, but which need to be applied from scratch with more professional programs. I’m going to help you get an idea for some fairly advanced EQ, Compression, and FX techniques that you can apply on your own, which should also help you understand fundamentally how to create a great mix.  All of the specific settings I’m recommending are my own personal preference – they are by no means the “standard” or “correct,” but should give you an idea of what you can go for when you mix.  This article – Part 1 – will focus on EQ.  Part 2 will focus on dynamic processing and time-based effects. Equalization (EQ) – The Golden Rule: Cut before you Boost Always cut before you boost.  Think about it this way – you want to cut out the nasty, keeping the sweet.  If you boost too much, you’re keeping the nasty, and adding a little sweet, but hearing mostly the nasty.  You’re raising the noise floor when you boost.  Any audio processor is going to color or change the tone and quality of whatever audio is fed through it.  Equalizers sound better if you attenuate frequencies rather than raising the gain. Search and Destroy: Find the bad frequencies before you cut them A quick way to find bad frequencies (whether it’s from the room acoustics, sibilance, the instrument, or an electronic issue) is to boost a specific frequency with a sharp Quality Factor* (8 to 10) very harshly (boost it 10dB) and listen. While you have that boosted, change the frequency and search around to find the offending frequency.  Once you’ve found it, try softening the Q (maybe 3 to 7), then gently cutting until you can’t hear that frequency standing out any more.  You could also reverse that and cut it 10 or 20 dB, then slowly raise it back up until you barely hear it.  You might use this technique to find a ringing in a snare drum – and you might want to make a note of the frequency (say it’s 185 Hz), then also check for the ringing at the harmonics of that same frequency (double or half the Hz, so check 92.5 Hz and 370 Hz). *The Quality Factor (Q) defines the sharpness of the band of frequencies affected by an equalizer.  A lower number widens the band while a higher number narrows the band. Picture 14 Kick – Cut 200 Hz with a fairly wide bell curve (with a Q of 1 to 3) to attenuate some of the “boxy” sound of the drum.  Boost 3.5 kHz to 5 kHz with a fairly sharp bell curve (with a Q of 3 to 6).  The phat nasty (the good kind of nasty) bumpin’ frequencies in the kick drum are around 80 – 150 Hz.  Different size kick drums are centered around different frequencies.  50 Hz and below is the subtle sub frequencies that add depth to the kick.  Don’t go boosting that low end just because you don’t think you can hear it well enough.  You don’t want to compromise that nice top end you’ve carved out.  The biggest mistake people make is mixing the low end too hot.  A bass-heavy recording will muddy up your entire mix, and sound like a puny little chipmunk squeaking through someone’s computer speakers that can’t reproduce anything below 400 Hz. Snare – Use a high pass filter around 100 – 150 Hz, because there are generally no frequencies generated by the snare drum below that threshold that you need.  Try gently cutting the mid-range (around 600 Hz) with a wide bell curve (Q of .5 to 2) to take out some of the harsher noises in this range that aren’t always frequencies most prominent in the snare drum.  Sometimes you still want to feel the weight of the snare, which you can do by leaving a little low end around 150 Hz to 300 Hz.  That crack you want in the snare is usually between 1 kHz and 4 kHz, depending on the snare drum.  Try boosting 3.5 kHz to get more crack…but don’t have a crack attack! *For that epic 70’s & 80’s snare sound, try scooping a big chunk of the mids out (600 Hz) leaving that low end and the crack. High Tom – Use a high pass filter around 50 Hz to 80 Hz, or even higher.  Toms are tricky – sometimes they sound great without any tweaking, other times it’s a much-needed fix.  Try cutting around 500 or 600 Hz with a soft Q (1 to 3) to take out some of the “nasty” box-like sound.  Boost that frequency first so you can hear what sort of “nasty” I’m talking about.  The slap/crack of the high tom might be around 3.5 or 4 kHz, so you can boost that with a sharp Q (similar to the slap of the kick) to help the top end cut through the mix.  The sweet spot in the low end is entirely dependant on the tom, but is usually around 150 to 300 Hz. Low Tom – Only use a high pass filter if the floor tom is too bassy, and then use it around 50 Hz.  Similar to the kick, cut around 200 or 250 Hz to attenuate that boxy sound, then subtly boost (if needed) 3.5 kHz to help the attack/slap cut through the mix. High Hat – Use a high pass filter starting anywhere from 200 Hz to 450 Hz.  You only really want the sweet, smooth, and bright frequencies between 800 Hz and 10 kHz.  If you’ve got a harsh sounding hat, try cutting 1 or 2 kHz with a medium Q (3-6). Overheads – If you’re just going for the cymbals, cut most of the other drums out by using a high pass filter around 200 Hz to 400 Hz.  Sometimes you’ll want to gently attenuate (cut) 600 Hz with a wide Q (.5 to 2) to soften those harsh mid frequencies not always favored in drums. Vocals – Use a high pass filter starting around 100 to 150 Hz.  Listen carefully, and depending on the singer, you might want more low end or less.  Cutting low end (200 Hz to 300 Hz) off a vocal is like cutting off the weighted chains holding it down beneath the ocean of sound and lifting it above the mix.  If you recorded the vocals in a small room, there could be “room modes” (frequencies that are more reverberant than others due to the dimensions of the room), that are affecting your vocal.  You can find these by using the search and destroy method and listening for frequencies that sound like they’re humming or feeding back.  You may want to cut a little harshness around 1 kHz to 2 kHz (with a medium Q between 4 and 6), which is where most of the harsh sibilance (s’s, t’s, v’s, f’s) is.  You don’t want to cut too much of this, because the sibilance helps us understand what the vocalist is saying – you just don’t want these frequencies to sound too harsh on the ears.  Attenuating the sibilance is usually best accomplished with a de-esser (a dynamic processor that will be covered in Part 2).  You can add “air” to vocals to make them sound lighter and brighter by boosting 5 Hz and higher with a shelf EQ…or a soft shelf boost after 1 kHz. Acoustic Guitar – In a full band mix, you want the acoustics to really pop out of the mix, and the best way to do this is to cut off all the baggage – so use a high pass filter as low as 100 Hz and up as high as 400 Hz (use your own judgment, all acoustics are different).  You should also look for frequencies in the lower mid-range that are excessively boomy or humming (most acoustic guitars have a natural resonance – a frequency at which they vibrate/reverberate the most).  Before you boost 3 kHz – 10 kHz to get that sparkly you want, try cutting the low end and mid range more with a wide EQ band. Electric Guitar – For rhythm guitars, you want to keep some of the warmth and power of the low end, but you still want to get rid of the unnecessary rumble deeper down.  Try using a high pass filter around 100 Hz.  For lead guitars, try using a high pass filter up to 150 Hz or 200 Hz.  Start low and raise the frequency until you hear it begin to affect the center of the tone, then back off (lower the frequency) to where you can’t tell the difference.  You might not be able to hear the lower frequencies you’re cutting out, but you are cleaning up your mix!  This allows more room for the kick and bass guitar in the low end without the guitars muddying up everything. You may have noticed I didn’t mention anything else besides the high pass filter when mixing electric guitars…  That’s because I believe it’s better to go for a great tone with the amp and mic placement than try to fix it with EQ.  I rarely use more than a high pass filter on my electric guitars. Piano – Another instrument similar to an acoustic guitar in a full band mix – you want it to pop above everything else, and be able to mix it fairly low and still have it stand out.  Use a soft, but generous high-pass filter between 100 Hz and 200 Hz.  To help the attack of the keys sit above your mix, try gently boosting anywhere between 1 kHz and 5 kHz. Bass – Your bass guitar should be married to your kick drum.  It should sound as though the kick drum is your bassist thumping his strings – the attack at the front of every hit.  To achieve this, you should find what frequency your kick is centered on and carve out that frequency in your bass guitar (cut about 5 dB and see what it sounds like) to make space for the kick.  To get rid of the boominess of a bass guitar, try cutting between 80 Hz and 150 Hz.  Cutting to carve out that space for the kick drum will help control a boomy bass as well.  If you want to add a little presence (brightness) to your bass, try boosting sharply on a frequency between 1kHz and 3 kHz. If you want to hear examples of songs I’ve mixed of various genres (if you don’t trust me), go to www.jakehartsfield.com and click on the DP logo. Your ears are your greatest tool.  Listen to a lot of music and try to imitate the sounds you hear when you’re mixing.  Play a song with a great drum track or lead vocal through your speakers, then immediately switch back to your mixing software and try to replicate that sound – try to hear and find how sharply they EQ’d the instrument, which frequencies stick out, and which sound the sweetest to your ear. Happy Hearing! Posted by TuneCore at 10:35 AM | Permalink TrackBack TrackBack URL for this entry: http://www.typepad.com/services/trackback/6a00d83455f72e69e20120a6211738970c

	Drum Mixing Techniques 1.) Snares love plate 'verb 2.) Multiple 'verbs on snare often works 3.) Cutting top end off room mics is often good 4.) "Modern" kick drums have a lot less low end than you may think, and a lot more 4-6K... you may end up boosting +12 db in that range... don't fret, it happens. 5.) Nail you kick/snare compression attack/release curves right away. Once you nail that (and you'll hear it when you do) you've got the drum sounds nailed. 6.) Don't pan the toms too far... 75-45% wide does the trick. 7.) Don't go insane with the top end of the OH's.... too much 10K+ ends up with a very amateur sound. 8.) Snare EQ: HP around 70hz, boost around 120-240, cut around 500-700, boost at 1.2K, and look for something between 3 and 10 K for more boost depending. Depends on what you got and want. Piccolo snares tend to like 6-7k boosts. 9.) Try parallel compression on only the rooms--run one fairly mild, and the other set to annihilate. Balance them out until it sounds cool. 10.) Knock out some 200 and 800 hz in room mics... leave the rest of it alone except for maybe some mild 8 Khz lowpassin'. If the kick has to be tight, HP the signal as well. 11.) Ruthlessly cut the lower mids on kick.... 6-15 db cuts should do around the 300-500 area. Set bandwidth to taste... the tighter the kick you want the more around 150-250 you should be rollin' off on. 12.) Pick either your OH's or rooms as being dominant. Don't put 'em in at the same levels--have one louder than the other. The "modern" way is to choose the rooms a bit more--to balance out the ultra compressed and loud direct mics. Most OH's these days are cymbal info and a little clarity only and are often low in the mix (like -12 db on the meters it seems). 13.) Limit *AND* compress kicks and snares. Love compression with a vengeance for that Lord Alge sound. 14.) Put a stereo widener on your OH's... makes the drumkit image bigger, can make the snare sound a bit fatter too. 15.) Put 20 ms delay on your room mics to get that Albini sound. 16.) Apply vigorous amounts of tape saturation as the first plugin in your chain.... you'll need less compression later on. Gets a good vintage/indie type sound if you lay it on there. 17.) 20-80 ms of predelay on snare 'verbs can be cool. 18.) Non-lin verb sounds on drums is probably going to come back in style--I've already gotten requests for "big 80's drum sounds, tons of reverb" from young bands. 19.) Don't compress your OH mics. 20.) See if you can get a good sound using only your OH's and (some) room mics. Add minor amounts of close kick/snare (maybe not snare) for a vintage type sounds. It can be interesting how great you can get this to sound (except forget it when doing "modern" rock or metal--you need all the close mics you can get). 21.) When in doubt use triggers/samples. However, if it's a "learning" session or your trying to improve your chops don't use those things until you learn how to mix without it. Use the bare minimum when you do use 'em... James Meeker. http://www.myspace.com/jamesmeekerproductions