general compression techniques - tips and help needed

[/////\\\\\\\////////]

i am after help on compression processes with sequenced, sample based music (in this case hip hop, with programmed samples drums, samples, sub bass, and some live instruments - genrally relatively few elements).

can anyone give genreal advice??

eg
- on compressing drum elements, is this normally done as one [ie all drums compressed together], or would you compress elements, such as kick and snare separetely.

- would you add some compression to all the elements in the track [eg samples and live instrumentation]

- how does bass relate to drums in compression

- what is a good starting point for compression ratios in sequenced music [i know ultimately it is what sounds best], especially on the drums

any help or links to articles is mad apreciated

cheers tom smile.gif

btw i'm using Sonar/atari

[wil milton]

Hello Tom,

What I find works for me is treating all tracks dependently of each other. What u do to one, efects the other. When u use a varaiable gain, such as a compressor, however, u do not want to compress the snare at the same level as the kick. The kick and the sanre have two different roles in a mix. The kick sets the pulse, and is suppose to cut right through the track. U can sidechain the bass and the kick drum together, due to them supporting each other. If u compress all of the drum tracks, u wind up with a really thin drum mix. Keep in mind that a compressor is to be utilized to balance out all levels. The only time u compress a mix entirely is when u go to mastering. which is multi-band compressing. On ratio, try 4.1 on vocal, bass, and drums. Anything higher than 9.1 is the opposite of compression, which is limiting. I hope this is helpful B blessed.

Peace Always
Wil MIlton

[ May 31, 2004, 10:11 AM: Message edited by: wil milton ]

[JorgeG]

Hello Tom Cristavao,

I couldn't find this link this article is from keyboard magazine 12/98 (title: 99 tips you can't live without)....

http://jukar.biostat.wisc.edu/pics/compression1.jpg
http://jukar.biostat.wisc.edu/pics/compression2.jpg

[BastiBoi]

EQ Secrets:

There are very few absolute "Rights and Wrongs" when it comes to EQ. Basically, if it sounds good to you, it's right. There are some generally accepted thoughts on the matter though, so we'll go over some of them as starting points. One generally accepted thought is that most vocal mics are very midrange heavy. Typically a cut in the midrange along with a slight boost of the bass and treble frequencies can compensate for this. Now remember, every voice is different, so don't just set every vocal mic in your system to one setting and go. You have to listen to the individual characteristics of each voice. If someone has a deep, booming voice you may find yourself cutting the bass and boosting the mids and highs. A female vocalist may have a very light, "airy" voice and may need some help (boost) in the low and mid areas. "Sweep" type or Parametric midrange controls are great because they let you "dial in" the particular problem frequency and then boost or cut it. For example, to find a problem frequency, you can turn the "level" control of the mid-sweep to full cut and then sweep the "frequency" knob until you find the exact frequency that you want to cut. Then re-adjust the "level" knob to the actual amount of cut you want. Fully parametric EQ's let you adjust the band-width as well. This is generally called a "Q" adjustment and it determines how big of a chunk (in octaves) will be cut or boosted. Typically these can be adjusted anywhere from 0.1 to a full octave. Most mixers don't have fully parametric EQ, while semi-parametric EQ's (no Q adjust) are quite common.


Another generally accepted thought is that there really isn't much below 40Hz that anyone wants to hear live. On a recording it may be a different story, but live you pretty much want to cut anything below 40Hz. This will do several things: it will reduce stage rumble, reduce that "flabby, booming" kick/bass sound, will increase headroom (which is the reserve power capability) in your power amplifiers (because now they won't have to reproduce those power robbing low frequencies), and generally clean up the whole mix. If your amp, crossover or mixer has sub-sonic filters... use them! Those filters are the quickest way to cut down on those unwanted low frequencies below 40 Hz. Also, on midrange and treble instruments (vocals, horns, guitars, keys, etc.) use the channel low cut (if available). These vary from 75Hz to 100Hz and give you all of the above mentioned benefits without adversely affecting the sound. Now I know that some 5 & 6 string bass players and keyboard players will insist that there is desirable musical content below 40Hz. However, there are harmonics and overtones that still let you hear the low "B" being played on a 5 string bass, you just won't hear the fundamental (approx. 30Hz). I think the trade-off (a clean, punchy mix) is worth it. You can decide for yourself.Part of getting the sound you want is knowing where an instrument lies in the frequency spectrum and part of it is just plain experimentation.


Most people know that a kick drum is a bass instrument right? Well, yes and no. Most of what you hear from a kick drum is in the bass region, true, but try boosting the kick at 1K. Hear that extra "snap" that you just added? Or try boosting the snare around 250Hz. That midrange/treble drum suddenly sounds huge and punchy. Bass guitar is often "mis EQ'd". Instead of pumping up the low end until the windows shake loose, try cutting around 400Hz to get rid of some "mud" and boosting around 1.5K to add some "twang". The bass will cut through the mix with great definition and still have plenty of low end. Some instruments (like a piano) cover a very wide range of the spectrum and can be very difficult to EQ. I've seen sound people use six EQs on a grand piano (4 channel mixer EQ, 2 outboard parametric). Of course this probably means it wasn't mic'ed properly to begin with. Proper mic selection and placement are necessary to get as close to the right sound as possible so that major EQ work is not needed (but that's another article). Electric or sampled pianos are pre-EQ'd and typically don't require as much work. Maybe cut the bass a bit and boost the treble so it fits in the mix better (keep in mind that a solo instrument should be treated differently and may not require as much EQ, if any, as an instrument that you want to fit into a mix). When trying to fit a lot of instruments into a mix, try what I call "contrary" EQ. If the instrument is in the bass range, try boosting the high-mid or treble frequencies, add some low end to a female vocal, or add some "edge" to a male vocalist with a treble boost.


Graphic EQ's are usually used on the overall mix and in the monitor mix. For the overall mix you would want to use a graphic EQ to fine tune your mix to a particular room. If it's a "dead" room (acoustic tile ceiling, thick carpet and full of people), you may want to boost the high end and/or cut some low frequencies, if it's a "live" (hard floors, walls, ceiling, etc.) room you may need to cut the high/mid and treble a bit. Typically, a 10 to 15 band graphic (2/3 to 1 octave) is sufficient for mains. For monitors you may need a 30 band (1/3 octave) graphic EQ for each monitor mix. This type of narrow-band fine tuning lets you precisely locate and cut frequencies that cause "feedback", rumble, hum, etc. I could write fifty more pages and not cover the half of it. Please remember that these are suggestions for starting points and they are my personal opinions. Experiment and find out what works best for you. Also remember there is no substitute for quality equipment. If you use a poor sounding mic, mixer, amp, instrument, etc., no amount of EQ will totally fix it.

graemlins/bolt.gif

[BastiBoi]

Guide To Compression & Limiting:

Settings and Characteristics

Threshold
Ratio
Hardknee
Attack
Autoattack/ Release
Holdtime
Stereo Link
All in the ear
Using compressors
Side Effects
De-Essing

Introduction

For the benefit of those who are still a little unsure as to what a compressor does, it simply reduces the difference between the loudest and quietest parts of a piece of music by automatically turning down the gain when the signal gets past a predetermined level. In this respect, it does a similar job to the human hand on the fader—but it reacts much faster and with greater precision, allowing it to bring excessive level deviations under control almost instantaneously. Unlike the human operator though, the compressor has no feel or intuition; it simply does what you set it up to do, which makes it very important that you understand what all the variable parameters do and how they affect the final sound. In order to react quickly enough, the compressor dispenses with the human ear and instead monitors the signal level by electronic means. A part of the circuit known as the 'side chain' follows the envelope of the signal, usually at the compressor's output, and, uses this to generate a control signal which is fed into the gain control circuit. When the output signal rises past an acceptable level, a control signal is generated and the gain is turned down. Figure 1 shows a simplified block diagram of a typical compressor circuit.

Threshold:

With manual gain riding, the level above which the signal becomes unacceptably loud is determined by the engineer's discretion: if it sounds too loud to him, he turns it down. In the case of a compressor, we have to 'tell' it when to intervene, and this level is known as the Threshold. In a conventional compressor, the Threshold is varied via a knob calibrated in dBs, and a gain reduction meter is usually included so we can see how much the gain is being modified. If the signal level falls short of the threshold, no processing takes place and the gain reduction meter reads OdB. Signals exceeding the Threshold are reduced in level, and the amount of reduction is shown on the meter. This means the signal peaks are no longer as loud as they were, so in order to compensate, a further stage of 'make-up' gain is added after compression, to restore or 'make up' any lost gain.

Ratio:

When the input signal exceeds the Threshold set by the operator, gain reduction is applied, but the actual amount of gain reduction depends on the 'Ratio' setting. You will see the Ratio expressed in the form 4:1 or similar, and the range of a typical Ratio control is variable from 1:1 (no gain reduction all) to infinity:1, which means that the output level is never allowed to rise above the Threshold setting. This latter condition is known as limiting, because the Threshold, in effect, sets a limit which the signal is not allowed to exceed. Ratio is based on dBs, so if a compression ratio of 3:1 is set, an input signal exceeding the Threshold by 3dB will cause only a 1 dB increase in level at the output. In practice, most compressors have sufficient Ratio range to allow them to function as both compressors and limiters, which is why they are sometimes known by both names. The relationship between Threshold and Ratio is shown in Figure 2, but if you're not comfortable with dBs or graphs, all you need to remember is that the larger the Ratio, the more gain reduction is applied to any signal exceeding the Threshold.


Hardknee:

This is not a control or parameter, but rather a characteristic of certain designs of compressor. With a conventional compressor, nothing happens until the signal reaches the Threshold, but as soon as it does, the full quota of gain reduction is thrown at it, as determined by the Ratio control setting. This is known as hard knee compression, because a graph of input gain against output gain will show a clear change in slope (a sharp angle) at the Threshold level, as is evident from Figure 2. Other types of compressor utilise a soft knee characteristic, where the gain reduction is brought in progressively over a range of 1OdB or so. What happens is that when the signal comes within 1OdB or so of the Threshold set by the user, the compressor starts to apply gain reduction, but with a very low Ratio setting, so there's very little effect. As the input level increases, the compression Ratio is automatically increased until at the Threshold level, the Ratio has increased to the amount set by the user on the Ratio control. This results in a gentler degree of control for signals that are hovering around the Threshold point, and the practical outcome is that the signal sounds less obviously processed. This attribute makes soft-knee models popular for processing complete mixes or other sounds that need subtle control. Hard knee compression can sometimes be heard working, and if a lot of gain reduction is being applied, they can sound quite heavy-handed. In some situations, it can make for an interesting sound—take Phil Collins' or Kate Bush's vocal sounds, for example. The dotted curve on the graph in Figure 2 shows a typical soft-knee characteristic.

Attack:

The attack time is how long a compressor takes to pull the gain down, once the input signal has reached or exceeded the Threshold level. With a fast attack setting, the signal is controlled almost immediately, whereas a slower attack time will allow the start of a transient or percussive sound to pass through unchanged, before the compressor gets its act together and does something about it. Creating a deliberate overshoot by setting an attack time of several milliseconds is a much-used way of enhancing the percussive characteristics of instruments such as guitars or drums. For most musical uses, an initial attack setting of between 1 and 20 mS is typical. However, when treating sound such as vocals, a fast attack time generally gives the best results, because it brings the level under control very quickly, producing a more natural sound. Release: The Release sets how long it takes for the compressor's gain to come back up to normal once the input signal has fallen back below the Threshold. If the release time is too fast, the signal level may 'pump'—in other words, you can hear the level of the signal going up and down. This is usually a bad thing, but again, it has its creative uses, especially in rock music. If the release time is too long, the gain may not have recovered by the time the next 'above Threshold' sound occurs. A good starting point for the release time is between 0.2 and 0.6 seconds.


Auto Attack/Release:

Some models of compressor have an Auto mode, which adjusts the attack and release characteristics during operation to suit the dynamics of the music being processed. In the case of complex mixes or vocals where the dynamics are constantly changing, the Auto mode may do a better job than fixed manual settings. Peak/RMS operation: Every compressor uses a circuit known as a side chain, and the side chain's job in life is to measure how big the signal is, so that it knows when it needs compressing. This information is then used to control the gain circuit, which may be based around a Voltage-controlled Amplifier (VCA), a Field Effect Transistor (FET) or even a valve. The compressor will behave differently, depending on whether the side chain responds to average signal levels or to absolute signal peaks. An RMS level detector works rather like the human ear, which pays less attention to short duration, loud sounds than to longer sounds of the same level. Though RMS offers the closest approximation to the way in which our ears respond to sound, many American engineers prefer to work with Peak, possibly because it provides a greater degree of control. And though RMS provides a very natural-sounding dynamic control, short signal peaks will get through unnoticed, even if a fast attack time is set, which means the engineer has less control over the absolute peak signal levels. This can be a problem when making digital recordings, as clipping is to be avoided at all costs. The difference between Peak and RMS sensing tends to show up most on music that contains percussive sounds, where the Peak type of compressor will more accurately track the peak levels of the individual drum beats. Another way to look at it is to say that the greater the difference between a signal's peak and average level, the more apparent the difference between RMS and peak compression/limiting will be. On a sustained pad sound with no peaks, there should be no appreciable difference. Peak sensing can sometimes sound over-controlled, unless the amount of compression used is slight. It's really down to personal choice, and all judgements should be based on listening tests.


Holdtime:

A compressor's side chain follows the envelope of the signal being fed into it, but if the attack and release times are set to their fastest positions, it is likely that the compressor will attempt to respond not to the envelope of the input signal but to individual cycles of the input waveform. This is particularly significant when the input signal is from a bass instrument, as the individual cycles are relatively long, compared to higher frequencies. If compression of the individual waveform cycles is allowed to occur, very bad distortion is audible, as the waveform itself gets reshaped by the compression process. We could simply increase the release time of the compressor so that it becomes too slow to react to individual cycles, but sometimes it's useful to be able to set a very fast release time. A better option is to use the Hold time control, if you have one. Hold introduces a slight delay before the release phase is initiated, which prevents the envelope shaper from going into release mode until the Hold time has elapsed. If the Hold time is set longer than the duration of a single cycle of the lowest audible frequency, the compressor will be forced to wait long enough for the next cycle to come along, thus avoiding distortion. A Hold time of 50ms will prevent this distortion mechanism causing problems down to 20Hz. If your compressor doesn't have a separate Hold time control, it may still have a built-in, preset amount of Hold time. A 50ms hold time isn't going to adversely affect any other aspect of the compressor's operation, and leaves the user with one less control to worry about.


Stereo Link:

When processing stereo signals, it is important that both channels are treated equally, for the stereo image will wander if one channel receives more compression than the other. For example, if a loud sound occurs only in the left channel, then the left channel gain will be reduced, and everything else present in the left channel will also be turned down in the mix. This will result in an apparent movement towards the right channel, which is not undergoing so much gain reduction. The Stereo Link switch of a dual-channel compressor simply forces both channels to work together, based either on an average of the two input signals, or whichever is the highest in level at any one time. Of course, both channels must be set up exactly the same for this to work properly, but that's taken care of by the compressor. When the two channels are switched to stereo, one set of controls usually becomes the master for both channels though some manufacturers opt for averaging the two channel's control settings, or for reacting to whichever channel's controls are set to the highest value.


All in the ear:

You may have noticed, or at least read about, the fact that different makes of compressor sound different. But if all they're really doing is changing level, shouldn't they all sound exactly the same? As we've already learned, part of the reason is related to the shape of the attack and release curves of the compressor, and of course peak sensing will produce different results to RMS, but at least as important is the way in which a compressor distorts the signal. Technically perhaps, the best compressor is one that doesn't add any distortion, but most engineers seem to like the 'warm' sound of the older valve designs which, on paper, are blighted by high distortion levels. The truth is that low levels of distortion have a profound effect on the way in which we perceive sound, which is the principle on which aural exciters work. A very small amount of even-harmonic distortion can tighten up bass sounds, while making the top end seem brighter and cleaner. The best-sounding contemporary compressor designs include valve models with a degree of distortion built in, while others use FETs, which mimic the behaviour of valve circuits. As digital recorders and mixers are introduced into the signal chain, more people are becoming interested in equipment that can put the warmth back into what they perceive as an over-clinical sound.


Using compressors:

One problem newcomers to recording seem to have is deciding where in their system to patch the compressor. A compressor is a processor rather than an effect, so it should be used via an insert point or be patched in-line with a line-level signal. If you have a system without insert points and you want to compress a mic input, you may be able to use your foldback (pre-fade send) in an unconventional way to get around the problem, as shown in Figure 3.

Here's how to do it: Plug the mic into a mixer channel, set the mic gain level as normal, but turn the channel fader completely down. Turn the pre-fade aux send control to around three-quarters up, and do the same with the pre-fade master control, if there is one. Turn the pre-fade send fully down on all the other channels. Now you can take your mic signal (now boosted to line level), from the pre-fade send output, feed it into the compressor and bring it back into another channel of the mixer this time into the line input. And there you have it: your compressed mic signal. Most engineers will normally add some compression to vocals while recording, and then add more if necessary while mixing. Working this way makes good use of the tape's dynamic range, while helping to prevent signal peaks from overloading the tape machine. It is best to use rather less compression than might ultimately be needed while recording, so that a little more can be added at the mixing stage if required. If too much compression is added at the beginning, there's little you can do to get rid of it afterwards. Similarly, if you have a compressor with a gate built-in, it might be better to leave this off when recording, and only use it while mixing. This will prevent a good take from being wrecked by an inappropriate gate setting. A further benefit of gating during the mix is that the gate will remove any tape hiss, along with the original recorded noise. If a gate is allowed to close too rapidly, it can chop off the ends of wanted sounds that have long decays, especially those with long reverb tails, so most gates (and expanders) fitted to compressors have either a switchable long/short release time, or a proper variable-release time control.


Stereo Link:

When processing stereo signals, it is important that both channels are treated equally, for the stereo image will wander if one channel receives more compression than the other. For example, if a loud sound occurs only in the left channel, then the left channel gain will be reduced, and everything else present in the left channel will also be turned down in the mix. This will result in an apparent movement towards the right channel, which is not undergoing so much gain reduction. The Stereo Link switch of a dual-channel compressor simply forces both channels to work together, based either on an average of the two input signals, or whichever is the highest in level at any one time. Of course, both channels must be set up exactly the same for this to work properly, but that's taken care of by the compressor. When the two channels are switched to stereo, one set of controls usually becomes the master for both channels though some manufacturers opt for averaging the two channel's control settings, or for reacting to whichever channel's controls are set to the highest value.


Side Effects:

Most of the sound energy in a typical piece of music occupies the low end of the audio spectrum, which is why your VU meters always seem to respond to the bass drum and bass guitar. High frequency sounds tend to be much lower in level and so rarely need compressing, but even so, high-frequency sounds in the mix are still brought down in level whenever the compressor reacts to loud bass sounds. For example, a quiet hi-hat occurring at the same time as a loud bass drum beat will be reduced in level. One technique to reduce the severity of this effect is to set a slightly longer attack time on the compressor, to allow the attack of the hi-hat to get through before the gain reduction occurs. This is only a partial solution, and if heavy compression is applied to a full mix, the overall sound can become dull, as the high-frequency detail is reduced in level. Going back to the subjective effect of subtle harmonic distortion for a moment, some compressor designs make use of harmonic distortion or dynamic equalisation to provide an increase in high-frequency level whenever heavy compression is taking place. This helps offset the dulling of high-frequency detail, and can make a great subjective difference, but it isn't a perfect solution. More elaborate compressors have been designed which split the signal into two or more frequency bands and compress these separately. This neatly avoids the bass end causing the high end to be needlessly compressed, but it can introduce other problems related to phase, unless the design is extremely well thought-out.


De-Essing:

Another side chain-related process is the de-essing of sibilant vocal sounds. Sibilance is sometimes evident when people pronounce the letters 's' or 't', and is really a high-pitched whistling caused by air passing around the teeth. If a parametric equaliser is inserted into the side-chain signal path of a compressor and tuned to boost the offending frequency, the compressor will apply more gain reduction when sibilance is present than at other times. Most sibilance occurs in the 5 to 1OkHz region of the audio spectrum, so if the equaliser is tuned to this frequency range and set to give around lOdB of boost, then in the selected frequency range, compression will occur 1OdB before it does in the rest of the audio spectrum. The equaliser should be set up by listening to the equaliser output, and then tuning the frequency control until the sibilant part of the input signal is strongest. Figure 4 shows how a compressor and equaliser may be used as a de-esser. Some compressors have a built-in sweep equaliser, to allow them to double as de-essers without the need for an external parametric equaliser.


I should stress that these are just to get you started the ideal settings vary from compressor to compressor, which is why I come up with slightly different figures every time I write on the subject. The more gain reduction is used, the higher the level of background noise, so never use more gain reduction than is necessary. Virtually all recorded pop music has a deliberately restricted dynamic range, to make it sound loud and powerful when played over the radio. The more a signal is compressed, the higher its average energy level. In addition to compressing the individual tracks during recording or mixing, the engineer may well have applied further compression to the overall mix. This can be very effective, but don't choke the life out of a mix by over-compressing it either. When it comes to individual tracks, it is pretty much routine to compress vocals, bass guitars, acoustic guitars and occasionally electric guitars, though overdriven guitar sounds tend to be self compressing anyway! The most important of these to get right is the lead vocal, because even modest dips in level can make the Iyrics difficult to hear over the backing. Sequenced instruments are less likely to need compression, because you can control the dynamics by manipulating the MIDI data in the sequencer. My own rule is to avoid compression (or any other form of treatment) unless it's absolutely necessary. Even with vocals, if somebody gives me a perfectly controlled vocal take, I wouldn't want to compress it just because compressing vocals is the done thing. Compression is a very valuable studio tool, but like all tools, it is just a means to an end not an end in itself.

[BastiBoi]

GIVING YOUR RECORDINGS A 'PRODUCED' SOUND:


Why is it that some perfectly well-recorded songs sound like demos, while others sound like top commercial tracks? Paul White investigates the mystery of the 'produced' sound.

One of the questions we hear most from Sound On Sound readers is "Why doesn't my music sound as 'produced' as the music I hear on commercial CDs?" I'm sure you won't be too surprised when I tell you that there isn't a single, simple answer. Some people assume that the superior equipment used in pro studios is the key, but although competent gear is required to do the job properly, you don't actually need anything esoteric. Even when it comes to recording vocals you don't have to use expensive high-end tube capacitor mics -- artists such as Phil Collins and Mick Jagger often use relatively inexpensive dynamic models because that's what works best for them. A few years ago, the drum sound was what gave away most demos, but now we have good drum machines, drum samples and sample loops, as well as real drums, to choose from.

The secret of a produced sound starts with the source material. It doesn't matter what you do to your recording afterwards if this isn't up to scratch. It almost goes without saying that good timing and good tuning are essential, but the choice of sounds and the way in which acoustic instruments and voices are recorded has a huge bearing on the perceived quality of the end result.


Vox Clever

If you record vocals in a small, untreated room, the chances are that the resulting sound will be boxy, so place your mic somewhere near the centre (but not exactly in the centre) of a larger room and put up improvised screens (sleeping bags, duvets, blankets and so on) where necessary to kill the reflections. Used in this way, virtually any respectable mic will give you good results providing you use a pop shield. You can also record acoustic guitars in the same environment.

Vocals invariably need compression, but what kind and how much? Listen to what you've recorded and try to establish how much variation there is in the vocal level. If you hear a lot of fluctuation it might be better to use a model of compressor that can pin down the level without changing the sound too much. The compressors that come as standard in Yamaha digital mixers are good for this, as you can really pile on the gain reduction without changing the sound too radically; there are also analogue models that can do the same. On the other hand, you may feel the vocals need thickening as well as levelling, in which case a compressor with a character of its own might be better suited to the job. Tube and 'opto' compressors generally produce the fattest sounds, and of course there are software plug-ins that emulate just about anything you can buy in a rackmount box.

The goal is to get the vocal sitting nicely with the backing track so that you don't feel the urge to turn it up or down in different parts of the song. Professional engineers may also spend some time fine-tuning vocal levels with their mixer automation systems, and if you use either a digital mixer or a computer-based recording system you can do the same.


Key Facts

Synth sounds must be chosen with care, because a lot of factory patches are designed to sound big and impressive for the benefit of those who choose their new instruments on the strength of 'preset cruising'! What sounds wonderful on its own might take up too much space in a mix so, if you don't want to edit the patch, try using EQ to trim off excess bass or high end. The EQ'd patch might sound odd in isolation, but it may well fit the track better. Another tip for those reluctant to get into heavy editing is to layer patches to get the desired result. For example, a deep bass sound mixed with a more percussive patch might help you produce a bass that you can hear as well as feel.

It's important not to over-orchestrate your arrangements, especially when you have fat synth pads and overdriven guitars occurring at the same time. The same is true of some treated drum loops, which can actually take up a lot of space. If in doubt, listen to some commercial mixes in a similar style to the track you're working with. You may be surprised at how little there is going on at any one time."One of the questions we hear most from Sound On Sound readers is 'Why doesn't my music sound as 'produced' as the music I hear on commercial CDs?'"

It may help if you get your sounds as close as possible to correct at source so you don't need to use a lot of EQ. Few budget mixers have the kind of EQ that works well when called upon to make major tonal changes, and often you'll find that the more you EQ, the harsher, boomier or less focused your mix becomes.


Reduced Reverb

Once you've created space in your mix, don't give it all away by filling every available gap with heavy reverb. As it happens, reverb is one area where a decent-quality unit really helps, especially if you use a lot of small-room or ambient reverbs. You don't have to spend a fortune: the excellent Lexicon MPX100 costs around £200, yet still offers the general feel of Lexicon's more expensive studio processors.

Bear in mind that heavy reverb tends to push a sound to the back of a mix, so if you want a vocal to appear up-front you should use a fairly bright reverb, with 80mS or so of pre-delay. Don't overdo the decay time, either, especially with up-tempo songs. Other effects should also be used carefully -- use an effect because the track needs it, not because you happen to have it! Dramatic effects can be made even more dramatic if you use them for short sections of a song rather than having them full-on all the way through, and delay effects often work best when the delay time is related to the tempo of the song.


Master The Situation

What many people don't realise is just how great a difference is made to commercial records at the mastering stage. Prior to mastering, you might be surprised at just how ordinary some mixes sound. Mastering often involves nothing more than compression, limiting and equalisation, but it has a dispro -----

Favourite Strings Guitars and basses can be a dead giveaway that a recording is not a commercial one if they are poorly recorded. Sticking a mic in front of an amp is probably still the best way to get a live-sounding recording of a performance, but if this is not feasible there are so many good recording preamps around now that there's little excuse for getting a thin or buzzy guitar sound. However, go easy on the overdrive, and consider using less overdrive but combining it with compression if you need sustain. Use a gate to keep your guitar tracks clear of unwanted noise, and also try to reduce clutter in the arrangement: where two guitars are playing essentially the same chords, for example, first decide whether both guitars are actually necessary. If they are, consider using different chord inversions for one of the parts, or even a capo. Incidentally, acoustic guitars almost always sound better miked than DI'd. Basses can actually be more difficult to record than guitars, because although they may sound great in isolation when DI'd via an active DI box and a compressor, they can still lack punch in the context of the overall mix. Again, consider miking the amp or using a guitar DI preamp so you can add just a little overdrive to warm up the sound. Compression will help keep the sound even and punchy. A good tip here is to make any necessary EQ adjustments when the rest of the track is playing, because then you'll be able to make the sound match the track. If you EQ the sound first it might sound great on its own, but could get completely lost when the other faders are brought up.

---portionate effect because of the quality of the equipment being used and the expertise of the person using it. Yes, this is one area where the equipment does make a huge difference, though with all-in-one mastering processors now available at prices project studio owners can afford, it is possible to get a professional sound at home providing you have good ears and accurate monitors.

A good equaliser doesn't just change the spectral balance of a sound: it also seems to lift information out of a mix. One popular mastering technique is to apply an overall boost of just one or two dBs at around 15kHz with a wide bandwidth setting. This is what people mean when they talk about 'air EQ', 'sheen' or 'gloss'. With a nice equaliser this boost will lift out high-end detail while at the same time pulling the vocals forward, but it shouldn't make the sound harsh or toppy. Similarly, adding a gentle dip at around 180-250Hz may help clarify a muddy lower mid-range, while a boost at 70-90Hz will firm up a weak bass end. It is vital to use a classy equaliser for this job, though -- a cheap one just won't deliver the necessary fairy dust! (And a good mastering equaliser probably costs more than many people's entire computer-based recording system.) I use an SPL Vitalizer on some of my mixes, as it replicates many of the EQ functions of a mastering processor, and if you don't have the money to buy a high-end equaliser I'd recommend one of the lower-cost versions of the Vitalizer as an easy-to-use alternative.

A very gentle overall compression of around 1.1:1 with a threshold of -30 to -40dB will make a mix sound more even and more powerful. However, multi-band mastering processors add a lot of flexibility in the area of compression, because they give you the opportunity to perform operations such as applying more compression to the bass end than to the rest of the mix. This helps firm up the bass end only, and any spectral imbalance caused by the different compression ratios can be restored by adjusting the levels of the various frequency bands at the compressor's output.

Mastering also tends to involve limiting, a process similar to compression (but with an infinitely high ratio) that controls just the tips of loud peaks. Applying a little limiting will often make it possible to increase the avera"The secret of a produced sound starts with the source material. It doesn't matter what you do to your recording afterwards if this isn't up to scratch." ge level of a mix by several dBs without any side effects becoming audible. If you're starting from a 20- or 24-bit master and you reduce to 16-bit right at the end of the process, this has the benefit of using the whole of the bit resolution of the CD format, which means less noise, less distortion and better low-level resolution. It also makes your CD sound as loud as the 'produced' commercial CDs in your collection. Use a limiter specifically designed for mastering (such as the Waves L1 plug-in or the limiter in your mastering processor) and don't over-limit, or you will start to hear the difference. Usually 4-5dB of limiting is all that's needed. A note on limiting: Any decision taken to limit or not to limit is a musical one. Some musical styles apply heavy limiting as part of the musical style’s “sound”, others don’t. Production requirements may suggest limiting is needed, for example preparing your music for broadcast might necessitate limiting in order to compensate for the radio’s smaller dynamic range. In our example we exaggerated in our limiting setting – The student should understand that limiting to produce a 5.2dB attenuation is a bit heavy. Normally we should watch out for a maximum of 4dB attenuation.

Processing via tube or simulated tube circuitry can also warm up a mix (which is why tube EQs and compressors are popular for mastering), but again you get even more flexibility if this tube processing comes as part of a multi-band package. For example, adding a little gentle tube saturation only to the low band will noticeably thicken the bass and kick drum without spilling over into the midrange and high end. Similarly, adding high-end saturation has an effect similar to an enhancer, enhancing detail and and lending gloss. The secret with all these treatments is to use them sparingly and always compare the processed sound with the unprocessed to make sure you have not gone too far. A good processor will transform a recording with just a dB or two of adjustment where needed. If you find you're using a lot of processing, suspect your basic mix of being too wide of the mark.


Summing Up

As you can see, the magic of musical production isn't something you 'paint' on at some point in the recording process, but is rather the result of attention to detail at all points throughout the recording, starting with the musical arrangement and choice of sounds. Nevertheless, processing at the mastering stage (ie. after your mix) can make a huge difference. Professional mastering is expensive for a reason: pro mastering engineers have great equipment and a lot of experience in using it. If you're not confident you have the necessary equipment and expertise to do your mix justice, think about getting your work professionally mastered, especially if it's destined for commercial release. If you're going to do this, don't do any processing at all on your final mixes -- leave each track just as it is.

On the other hand, if your mix is 95 percent there and you don't have the budget for pro mastering, don't be deterred from doing the job yourself, as there are now several hardware mastering processors (as well as innumerable software plug-ins) within the reach of serious project studio owners, and these can really help to get the job done.


graemlins/sleep2.gif

[/////\\\\\\\////////]

thanks for the tips and articles keep 'em coming - especially interested on people's own techniques and experiences graemlins/thumbsup.gif

[Dav]

Great Info.

I just printed off this whole thread...this will be my bedtime reading material for the next week...Thanks!

smile.gif

[drilla]

this is such a great thread.

thank you very much.

respect!

[SPINSTERWUN]

goto HTTP://WWW.THEPROJECTSTUDIOHANDBOOK.COM then click on COMPRESSION.

There are at least fifty (50) links on the explanation of compression.

Peace...

spin

[SPINSTERWUN]

This link will help you....

HTTP://WWW.COMPUTERMUSIC.CO.UK/TUTORIAL/EFFECTS1/EFFECTS1.ASP

[dennis f]

Vocals....

For recording vocals into our DAW we find that using a ratio of anywhere from 2:1 to 4:1 with a very high threshold just to tame the screamers a bit ....works well.

On mix down...6:1 is the one. Using soft knee, a very quick attack and leaving the compressor's release on AUTO. If your compressor has auto gain..use it. Also make sure you kind of leave the bandwith triggers alone, if you have them, unless you're really experienced using them.

For Kicks....I rarely do it as most of the processed kicks are already compressed to death.
There's really no guidline for this one ...just use you ears. Just make sure you don't squash it too much, unless that the desired effect, as it gets a bit poofy and spongey when you do this.

Most instruments you can get away with 4:1 a fast attack and auto release. Just watch your threshold levels. If you give it too much you'll squeeze the dynamics right out of the recording.

Note* After compression you definitely need to do a little eq'ing as all compression leaves colorization and dips some of the frequencies a bit. So you need to do a bit of a touch up job afterwards.

If you're using plugins...then check out Sonalksis
they have a compressor that has to be one of the best I've heard in a while..the eq is amazing also. The guys running this company used to work on Neve consoles. Man ....the compressor is sooooo smooth....sometimes ya can't even tell it's workin'.

peace
dennis

[michael]

Here's an entirely different take on the use of compression in pop music:

http://www.deepcity.org/temp/ohmforce_compression_article.JPG

[michael]

I forgot to mention that that is from the www.ohmforce.com (http://www.ohmforce.com) website (audio plug-in developers). They don't have it online anymore... maybe they realized it wasn't helping to sell any copies of their compression plug-in.