Mixing Tips

Digital Recording Tips

To avoid squashing, if it doesn't sound loud enough to your ears, turn up the monitor! If you find that you've been forced to apply limiting or compression just to keep the meters from overloading, then you've been going about this backwards. Instead, turn down your individual mix levels several dB, then get rid of any compression you were using to "protect" the 2-mix. Now your mix is at a lowered meter level, so turn up your monitor gain to arrive at the same loudness--only this time it won't sound squashed. Leave the monitor at that position as you continue to mix (mark it so you can get back to it).

In 24-bit recording you can make a perfectly good mix that peaks between -3 and -10 dBFS with no loss of quality, in fact, with improved quality. So if the mix gets too loud by your ears, then turn down the elements that are too hot in the mix instead of turning down the monitor again, with no fear of mixing "too low". In other words, a high monitor gain gives you less temptation to overcompress. High monitor gain does not necessarily mean high monitor output from the speakers--it means that the mix level had to be lower. For example, visit the CD Honor Roll and check out the great-sounding Lyle Lovett selection, which is close to the dynamics of a raw mix. Notice that in order to listen to it, you have to turn up your monitor gain. That's
approximately where your monitor control for a dynamic raw mix should be sitting (within 4-6 dB) before mastering. Obviously, a lot of today's hypercompressed masters would require turning down the monitor, but we're trying to show you how not to ruin the record in the mix stage (and hopefully not in the mastering,either!).

Know Your Monitors
But even when you do, never be fooled. Take your mixes around and listen to them on several other systems that you know; then go back into the control room and if they do not translate, try to adjust your mixes in the areas where they do not translate. HOWEVER, be aware of the extremes. If it sounds reasonably good in a car, for example, don't be tempted to turn up the highs for the car or it will screech (horribly) EVERYWHERE else. First of all, in the mastering we have much more experience in knowing how far to go and make sure that a recording is not made bass-shy just because it sounds boomy in a naturally-boomy car, for example.

Always Mix to the Highest Possible Wordlength
Even if the source tracks are 16-bit! Do not sample rate convert. When you're ready to bounce or prepare files, please see our guidelines page for suggestions on making file names and file types.

Track Important Instruments in Stereo
In the days of 8 track you had to be very careful about allocating tracks. But those days are gone. You have enough tracks to splurge now! So there's no reason to conserve on tracks during the tracking stage. The stereo image and depth of your final product will be determined by your skill in mixdown at using delays, reverberation, effects, and your skills in tracking, how you tracked your instruments. Try to make a plan beforehand of how your soundstage might look, where the instruments might be placed. Realize that it probably will not hurt, and probably will help to record your important instruments in stereo.

For example, even a pair of bongos that are destined to be on the right side of the soundstage will sound better if one bongo mike is panned full right and the other somewhat right of center. This is because the ear decodes the natural space and delays picked up by those microphones, actually enhancing their definition in the mix (if the room acoustics are good).

Another example: Electric guitar. Capture the direct to one track. Capture the output of the loudspeaker with a close mike to another track. Capture the medium distant sound of the speaker bouncing from the walls of the room with another mike. Listen to the combination of these sources panned to different places, and also listen in mono to make sure you have not created phase cancellations. By using stereo miking and natural room acoustics in the tracking, and possibly artificial delays and good stereophonic reverberation in the mixing, your mix will sound richer and deeper. Not everything should be tracked in stereo, but don't skimp on elements that will increase the depth and space of your recording. Of course you will need a
foreground, middleground, and background in the mix, but it's a lot harder to create a location and space for an instrument if you had only recorded it in mono.

In the mixing, use artificial reverberators that enhance depth and space and do not sound flat, plastic or "cheesy." Use artificial delays to locate instruments in space, not just simple panners.

Levels
Try to not exceed -3 dBFS peak on a peak meter on the highest peak of the mix. Low levels are perfectly acceptable in a 24 bit system. Once you see that the highest peak is in the range of, say, -10 dBFS to -3 dBFS, then from that point on, if you can hear it, the low level passages are ok.

Preserve Dynamic Range!
Assume that if anyone is going to ruin the master, let it be me (the mastering engineer). If the mix sounds good, then soft passages automatically are NOT too soft. Of course, if you think a soft passage sounds too soft in the mixing, then of course try to fix that during the mixing. But these can easily be dealt with and often more efficiently in mastering, as we have the context of the album in mind. If you have a VU meter, use it. With a sine wave, adjust it so 1 kHz, 0 VU is equal to -20 dBFS on the peakmeter. Use the VU, ignore the peak, and you'll start making better mixes.

Vocal Levels
Do make a lead vocal up (1/2 or 1 dB, you be the best judge) version. Do it NOW before you forget. It's a lot easier to do it NOW than to discover in the mastering that you should have. Occasionally do a vocal down (1/2 or 1 dB) version if you think it may be useful; then again, it only takes 3 minutes to do a vocal alt version when you're in the heat of mixing, but it takes forever to try to fix it in the mastering if you forgot.

Original Sources
If at all possible, deliver a generation that is as close to the original as possible. If it's on CD ROM, then cut a CDR directly from your hard disc files. Speed of cutting? Try to use Taiyo Yuden or other reputable blanks, and cut at 4X to 8X speed. These will PROBABLY produce the best results. Murphy's Law: Allow for Murphy. Do not ASSUME that all the files will transfer successfully over here and that the CD-ROMs you have cut are perfect. Allow for the possibility that on the very last minute of the very last hour of the very last day, we may have to go to a backup CD-ROM, or you may have to cut another, because of some error or other problem in the transfer. Do not paint yourself into a corner. Make backups. Do not destroy or erase any source hard disc at the origination studio until the mastering has been completed.

When, Why, and How to Make Stems
I've definitely reached the conclusion that the less compromise you can make in the mastering process, the better the result. Let's say you have an otherwise great mix, but which has too little bass instrument, too much kick, and the lower midrange is a little bit muddy. This is a potentially bad (not lethal) combination for mastering and if the client has time, I recommend a remix.
However, in situations like the afore mentioned, when time is tight, I have also asked the client for stems, and the results have ALWAYS been better than if I had mastered from the combined two-track. Next the question comes of whether to remix the stems without mastering processing in line or to try to mix/master in the same path. If it were a 40 track mix, I'd definitely mix first, then master, but with 3 to 6 stereo stems, I find that I can get the best results mixing and mastering at the same time; the result produces the best results and the least compromise. For example, the mastering processing is going to affect the clarity of the midrange and through "slop" will probably leak down into the bass region, hopefully for the better. But in the case of this lopsided mix I just cited, the mastering processing could easily make one range better while
making the other worse.

So, if mixing without the mastering processing, I may even try to take that into account, but if mixing with the mastering processing in place, I have it all in context at one time in the ideal acoustic of the mastering room.

Is this heresy? It's certainly a dangerous technique if placed in the wrong hands. You can end up with a less than ideal mix or less than ideal master if the mastering engineer does not think holistically. But if placed in the hands of an experienced mastering engineer, I think mixing from stems while mastering can produce the very best product. Separating out the bass instrument into a stereo stem with otherwise a mix minus, and sometimes separating out the vocal the same way can reduce the number of calls for a remix, I am convinced.

In other words, the lines between mixing and mastering have never been black and white. There has always been a gray area, and this method of mastering from stems grays it out even more!

The main purpose in this discussion was in the context of suggesting having a separate mastering engineer do the mastering from the stems, not in having a complete mixdown/mastering in one step.

If I am asked to both mix and master a project; if I am fortunate enough to do the mix from scratch in the mastering environment, then I probably would mix direct to 2 track without stems. I might run stems as a safety only; an ounce of prevention is worth a pound of cure. On that note, I note that with digital technology, a single 10 second mistake can cost a whole day of makeup! We don't need no stinkin' backups :-(
But I digress...

So, if I were mixing in the mastering environment I would probably just mix to 2 track WITHOUT MASTERING PROCESSING. But if I were mixing in a typical mixing environment, I would try to mix to stems if possible WITHOUT MASTERING PROCESSING.
What I'm saying is that although there is a gray area between mixing and mastering I don't advocate trying to combine the two processes when mixing completely from scratch. I only say that it is possible to do a good (better) job if you are the independent mastering engineer on the project and you receive stems instead of full mixes.

This article is taken from :http://www.digido.com/bob-katz/mixing-tips-and-tricks.html

EQ and Filter

There are five main types of filter: low-pass, high-pass, band-pass, band-stop and notch.

A low-pass filter allows low frequencies to pass and attenuates (reduces) high frequencies.

A high-pass filter allows high frequencies to pass and attenuates low frequencies.

A band-pass filter allows mid-range frequencies to pass and attenuates low and high frequencies.

A band-stop filter allows low and high frequencies to pass and attenuates mid-range frequencies.

A notch filter is a band-stop filter that covers a very narrow range of frequencies.

Most commonly found are the low-pass and high-pass filters. Let's look at the low-pass filter because this is used in sound engineering and widely in subtractive synthesis too...

The low-pass filter has a 'cut-off frequency'. Below this frequency, everything is allowed through unaltered. This is called the 'pass band'. Above this frequency, the signal is progressively attenuated at higher and higher frequencies. This is called the 'stop band'.

Actually at the cut-off frequency, the signal is 3 decibels lower in level than frequencies in the pass band. Clearly there is a gradual transition between the pass band and the stop band.

In the stop band, a filter is said to have a 'slope'. A low pass filter doesn't cut off high frequencies completely. Above the cut-off frequency it attenuates them more and more as the frequency gets higher.

At a certain point above the cut-off frequency, the degree of attenuation will be 6 dB (for instance). An octave higher in frequency, the attenuation may be 12 dB.

We would say that this filter has a slope of 6 dB/octave, an octave being a doubling of frequency.

It is common to find filters with slopes of 6, 12, 18 and 24 dB/octave. Clearly, a steeper slope means a more pronounced filtering effect.

A typical low-pass filter will have a control for cut-off frequency, and may have a switch for slope.

Now for equalizers...

Equalizers come in three main types: low frequency, midrange, high frequency. There are more options and subdivisions, but I don't want to get too complicated here.

Let's look at a low-frequency (LF) EQ and see how it differs from a low-pass filter...

A well-specified LF EQ will have the following controls:

FrequencyGainBell/shelf

The frequency control sets the frequency at which the EQ will start to take effect. It will operate on a range of frequencies lower than this.

Gain sets the amount of cut or boost to be applied. Usually there is up to plus or minus 12 to 18 dB.

If the LF EQ is set to 'shelf' then as the frequency of the signal drops below that set by the frequency control, then the amount of cut or boost will increase, until it reaches a maximum value. Beyond that, it will stay the same.

If the LF EQ is set to 'bell' then its the same as above, except that the amount of cut or boost will return to zero at lower frequencies.

LF and HF EQ sections do have a feature very similar to slope, but you don't get to control it. Some say that this is the factor that makes one EQ more 'musical' than another.

How To Record Vocals

The most common mistake is recording vocals too loud or too soft. The main goal to recording a solid vocal is to get all of the performance. It's not easy to set levels with a good, dynamic vocalist. As soon as you think you have the level pegged, they do something like move a few inches and you find out they are louder than you thought and meters are in the red. So you lower the level and find out that the meters are barely moving at all. If the vocalist is nervous and moving around, you might spend hours and never find an optimum level. The human voice is extremely dynamic, from soft whispers to piercing screams. If the level is too low, you will be bringing in noise and hum if you amplify it later. However, if you record too loud, there will be times when the file goes "over" which will likely result in damage that cannot be corrected later. The solution to this madness is to use a compressor in the chain after the preamp. The compressor, essentially, automatically lowers the volume when the input exceeds a certain threshold. It's like an invisible hand on a volume control. This allows a vocalist to get louder without going into the red. One of my favorite settings is to have the input to the compressor boosted so that all the "soft" words come through with a strong level. As soon as the vocalist gets louder, the clamping down begins and if they scream, it clamps down hard. The ideal is to have more consistent loudness no matter what they are doing.

Compressor and Noise in Recording

Compressor is an essential part in recording. Todays recordings always use compression in some individual tracks, and also possibly the entire mix.
But whenever you compress, you add noise. It always happens because there is no such noise free recordings. So here we have a problem that requires a solution if our recordings are not to suffer.
First of all we should know why compressor add noise? Let's look in more detail...
The action of a compressor is to reduce high levels in a signal so that they are closer to the lower levels. We say that the compressor reduces the dynamic range.
The problem now is that the signal now sounds quieter because the peak levels are lower. So we need to amplify the signal back up. This is called make up gain.
So now we have a signal that is as loud as it was before in the peaks, and the lower-level sections are louder too.So in bringing up the lower levels through make-up gain, the noise level is brought up too. What can be done?
In an individual track of just one instrument or vocal, while the instrument is playing the noise will be obscured, or masked as we call it. The noise will only be audible in the gaps when the instrument stops playing. If we can silence those gaps, then the noise will be inaudible. To do that, we need another piece of equipment - the noise gate.
A noise gate works by detecting the difference in level between the wanted signal and the noise. Where there is wanted signal, which will be higher in level than the noise, the gate lets it through. When there is just noise, the gate shuts it off. Noise gates can be tricky to set up, but the same effect can be achieved by simply muting sections of a track when the instrument isn't playing.
It is important to realize that the above only works where there are gaps. If you compress the entire stereo mix, then there will be no gaps, hence the noise gate cannot provide any benefit.

Recording Vocal

Microphones

Both condenser and dynamic microphones can be used to record vocals. Generally in professional studios, large diaphragm condensers are used, as they have a refined sound with a wide dynamic range and extended frequency response. Many excellent vocals, however, have been recorded on commonly available dynamics like the Shure SM58.
Choice of mic is down to what you have available, but in a situation where you have several different models, make your choice based on which mic suits the singer's voice for a particular song. Many engineers and producers will put up several mics initially to check which one sounds best.
Having chosen a microphone, it's preferable to mount it on a stand. Most mics come supplied with a mount, and the more expensive ones will have a suspended cradle mounting to isolate the microphone from shock and vibration.
It is possible to record vocals using a hand-held mic like an SM58 but, in terms of the sound being recorded, there are several reasons why this is not ideal. Firstly, there is handling noise to consider - the sound of the singer moving their grip on the microphone and moving it around will be picked up. Secondly, unless the singer is very experienced with mic technique, the mic will be held at different distances from the mouth at various times, resulting in small changes in timbre and level.

The next thing to consider about a microphone is its polar pattern. Many mics, especially less expensive ones, have a fixed polar pattern, usually cardioid. Others have a switchable polar pattern, but unless you are after a certain effect, switching it to cardioid is preferable.
Cardioid mics are the norm for recording vocals as they accept sound from directly in front and reject much of what comes from the back and sides. This is important in the context of the room where the vocal recording is to take place, as reflections from the walls may be picked up by the microphone, adding some of the ambient sound of that room to the vocal sound. This would obviously be more pronounced if an omni pattern was selected on the mic.

In most cases it is probably best to record a vocal in a dead-sounding area and add any ambience at the mixing stage using a reverb unit, because once ambience is recorded with a vocal, you are stuck with it.

6" is Ideal
A singer's distance from the mic can make a lot of difference to the sound recorded. A distance of 6" or so is perhaps a good starting point, although experienced singers will work the mic by leaning into it for some passages and moving back for louder sections.
Sing too far away from the mic and more of the room ambience will be picked up; sing closer to the mic and more of the proximity effect comes into play. Proximity effect is a pronounced boost in low frequencies which results in the voice sounding bassier when singing very close to the mic, and it can be successfully exploited by an experienced vocalist.
It is best to try to keep a vocalist at a consistent distance from the mic, particularly when doing multiple takes and where he/she has to leave the booth to listen to playbacks and then go back and sing the odd line. If the same distance from the mic is maintained, variations in volume and timbre between takes is minimized, and dropped-in lines will sound more natural.
Once a singer is at the optimum distance from the microphone, mark the position of their feet on the floor with gaffa tape so that they can go back to the same position each time, and don't forget to mark the position of the mic stand at the same time in case it is accidentally moved.
The height of the microphone on its stand in relation to the singer is also a factor to take into consideration. Some like to sing up to a mic suspended a little higher than them, but this can strain the voice if the head, neck and shoulders are stretched up. A mic that is suspended too low is also not ideal if it causes the singer to hunch over, although this at least puts less strain on the neck and shoulders.
An advisable starting position is to have the capsule level with the singer's mouth and then move it if necessary to suit the singer's most comfortable stance. Having the capsule level with the singer's mouth creates its own problems, as it is more susceptible to blasts of air, but there are methods to counter this, the most important of which is the use of a pop shield.
A pop shield is generally put up a couple of inches in front of the mic and its basic function is to stop plosives, which are the popping sounds from blasts of air usually produced by singing the vowels 'B' and 'P'. The pop shield also serves to protect microphones from spit and moisture produced by the singer.
Commercially available pop shields, which usually have a goose neck and a clamp allowing direct fixing to the mic stand, are fairly expensive.
If you cannot attach the pop shield directly to the mic stand, try using a second mic stand purely as support for the pop shield. If popping problems still persist, try getting the singer to sing
slightly to the side, above or below the mic. If a singer has difficulty doing this and needs to focus directly on the mic, put up another mic that is not plugged in and site it right next to the real vocal mic. Then let the singer sing into this dummy mic.

Whisper to a Scream
Microphones have to be connected to a pre-amplifier, and there are two options available. Connection can either be into the mic amp in a mixing desk's input channel, or into a standalone mic pre-amp. These provide a higher quality signal path to the recording medium than that provided by the average mixing desk.

Compression is near-essential to even out the performance when recording vocals. The human voice has a huge dynamic range (from a whisper to a scream, to use the old cliché), and a compressor will 'squash' that range a little. Don't go over the top, though; reducing the peaks by a few dB ought to be sufficient. Once compression is recorded you can't take it off, so it's best to err on the side of caution. More compression can, of course, be added as needed at the mixing stage.
Several of the standalone pre-amps on the market have their own compressor built in. When recording through a desk's input channel, a compressor should be connected via the channel's insert points.
EQ can also be applied when recording vocals, perhaps to remove a bit of nasal honk from a voice, to brighten up the sound a little, or, most usefully, to filter out some of the bottom end of the spectrum. Real low frequency sounds, such as outside traffic rumble or the sound of the singer's feet moving on the floor can be transmitted up the stand to the microphone, and an increase in bass due to the previously mentioned proximity effect can also be a problem.
To get around this, switch in a high-pass or bass roll-off filter. Most mic pre-amps and desk channels, and some mics, will have a switchable filter operating at somewhere between 75Hz and 100Hz, cutting out most of the low end below that figure.

EQ should, however be applied with caution. Adding too much top-end boost, for example, can often exaggerate the sibilance of the voice. It's best to record a vocal flat, but if you feel the need for EQ, use it sparingly. And while you may be tempted to use a noise gate or downward expander to cut out noise between phrases, our advice is not to. It's too easy to chop the end off notes and make the vocal sound unnatural.
Processing of this sort should be left to the mix stage, when time can be taken to set it up accurately.

Condenser or Dynamic?
Although there are other designs, the microphones most commonly used in studios today fall into one of two categories - condenser or dynamic. A microphone is simply a device which converts acoustic energy (sound waves) into electrical energy, and the dynamic and the condenser each do that in their own way. This has consequences for the sound produced, and hence the use to which each is put.
A condenser mic, also known as a capacitor mic, has a thin diaphragm that is supported around its rim at a small distance from a thicker backplate. The theory is that the two form the two electrodes of a simple capacitor, and are oppositely charged by the application of a polarising voltage. When the diaphragm moves in response to sound waves, the spacing of the diaphragm and backplate (and hence the capacitance) will vary, and this is used to generate the output voltage.
Because a voltage has to be supplied to the backplate and diaphragm, a mic of this nature needs a power supply.
This usually comes in the form of 48V phantom power supplied from the mixing desk or mic pre-amp. Condenser mics are more difficult to manufacture than dynamics and are therefore more expensive; they are also not as rugged and are more susceptible to changes in atmospheric conditions, so should be stored, and handled, with care.
In use, a condenser is generally more sensitive than a dynamic and has a better transient response. It also has a wider frequency response, so can pick up more top end than a dynamic, making it very useful for instruments like cymbals, acoustic guitars, and vocals.
Condensers can be built with two diaphragms, and by changing the voltage of the second diaphragm in relation to the first, the mic is capable of several different polar patterns - from omni-directional through cardioid to figure-of - eight.
Some condensers are designed with a valve in the circuitry; these do not need phantom power as they usually come with their own power supply. Valve mics provide a different tonality than the standard condenser, with an added warmth in the sound.
Another variation from the standard condenser design is the electret mic, which uses a permanently charged electret material to charge the capsule. These mics are usually cheaper than condensers and can often be run from a battery if you do not have a phantom power source.
Dynamic mics work because of the electromagnetic interaction between the field of a magnet and a moving coil conductor. A coil of wire, surrounded by magnets, is fixed to the back of the diaphragm, the motion of which results in the coil cutting through the magnetic field, inducing an electric current in the coil.
Unlike condensers, dynamic mics do not require any power supply. They are more robust, and can cope with high sound pressure levels.
Because dynamics are pressure- operated, their polar response can only be either omnidirectional or cardioid, and most handheld dynamic vocal mics are cardioids. Dynamics are also limited in their high frequency response, some having an upper limit of 16k (a good capacitor will go up to 20k).
Mics designed for stage use will often have a bass end roll-off built in to counteract the proximity effect, and many have a presence peak built into their frequency response somewhere up around 5k. This is designed to help vocals cut through a mix. Some very well-known rock singers record their vocals with dynamic mics for that particular punchy sound.

Microphone Polar Patterns
A cardioid (or unidirectional) mic is so named because of its heart-shaped response. It will pick up sound mostly from the front. Dynamic cardioid microphones are popular for vocals because of their off-axis exclusion, and robustness, but condenser cardioids are much better for the studio vocalist.
A figure-of-eight microphone picks up sound from both front and rear of the diaphragm, but because the opposite sides are out of phase, side-on sources get cancelled out. Figure-of-eight microphones have the potential for very accurate recordings.
A circle is the polar response of an 'ideal' omni-directional microphone. In practice, the response favours the 'open' side of the capsule at higher frequencies, so off-axis sources can be dull. Omni mics are particularly resistant to wind and handling noise.

Compression

Technical Reasons to Avoid Overall Compression on Your Album.
Save decisions on overall compression and individual tune equalization for an expert CD mastering house because:
1) The mastering house will have a more appropriate compressor with the proper attack, ratio, and release times exactly right for your music. If you mixed to digital tape, they will probably use a 24-bit digital compressor for the purpose.
2) They will likely be more experienced than you about the compromises, advantages and disadvantages of applying overall compression.
3) The mastering house can program that compressor with precision, adjusting it optimally for each tune in question. You're working out of context (without having the perspective of the entire album) by attempting these sorts of decisions during mixing.
4) The mastering house will be able to monitor your "CD in the making" using a calibrated monitoring system so that they know exactly how loud your "CD in the making" is compared to other CDs of similar music.
5) A good mastering house will be able to do all of this in a non-destructive, non-cumulative manner. In other words, after making a reference CD, they will be able to undo anything you are unhappy with, whether it be compression, EQ or levels. Whereas, most digital audio editing stations can only perform destructive EQ or compression, only with 16-bit word length, with a consequent loss of resolution as long internal words are either dithered (resulting in a veil if further processed), rounded (slightly better than truncated), or truncated to 16 bit.
6) For the same technical reasons, it is not a good idea to use a digital compressor (or any digital
processor) on your material before sending it for mastering. If you do feel the need to insert one of these boxes, for example, to give a demo CD to your client, be sure to also make a non-processed version to prepare for the mastering house. It is likely that the mastering house will have a fresher-sounding, more effective approach at polishing your material, and it's self-defeating if they have to try to undo what was done.
7) If you apply overall compression to your music, and your choice of compressor was wrong (e.g., the compressor you chose caused subtle pumping or breathing, loss of transients, loss of life or liveliness, etc. These are typical symptoms of "compressor misuse" on tapes I have received), the mastering house will have a difficult or impossible time attempting to undo the damage. As I've mentioned, mastering is like whittling soap; it is hard to undo compression.
This article is taken from http://www.digido.com/bob-katz/compression.html

15 Reverb TIPS

Reverb is essential in mixing, here are several tips in using reverb differently.

1. Rather than trying to make everything in the mix in the same acoustic environment, why not use a couple of really diverse reverbs to add some strange depth to your tunes? A really dry, upfront vocal works nicely alongside a really 'drowned' string section or a small bright room setting on the drums.
2. Try automating return levels if you have a digital mixer so that the reverb comes and goes in different sections of the song. By tweaking the aux send levels, manually, during the mix you can add splashes of reverb on the fly to add interest to snares or vocal parts.
3. Spend some time choosing or trying out different 'verbs. Different songs lend themselves towards different types and sounds. Don't just settle with what sounds good in solo...
4. Remember you can always EQ the send. Most large consoles offer you a choice of high and low EQ on the aux sends. On small desks, route the instrument / voice to another channel via a group or aux send, float this from the mix and send this to the reverb effect. Now you can add EQ to the send and even automate it as it's now on a fader. This is commonly used for those delays and reverbs that you want to move easily during the mix, such as wetter vocal in the chorus.
5. Reverse reverb is an old trick, where you can hear a vocal before a singer comes in, or a snare before it plays, easily using tape as you simply turn the tape over and record it backwards. You can do it using a computer, but you will have to move the audio to the right place after recording it.
6. A combination of reverbs on things can be good. A short setting for the snap sound with a longer bright plate can turn a biscuit-sounding snare into a more live sound.
7. In the old days it used to be called delay to plate. You sent the signal to a loop of tape then sent that to the reverb. The speed of the tape would adjust the delay as the time it took to get from the record head to the playback head. This gives, say, a voice a dry sound before the reverb comes in, giving a more upfront sound while keeping the wetness, which would usually take it to the back of a hall somewhere! Some people still use the tape method today for that old school sound.
8. Early reflections on drums can also give more of a tail or decay.
9. A nice gated verb on guitars to old spring verbs on snares or even the mighty space echo can sound unique when balanced in the mix. That will give you more distance and room for placing things in a mix, while adding that extra sparkle to the sound.
10. Reverse your sample, add reverb, then reverse your sample complete with reverb back around the right way again. This way, the reverb trail leads up into the sample, instead of trailing away from it.
11. For a different angle on the same reversed reverb theme, have the reverb trail panned left on a separate track, then the original sample centre-stage (ie. mono), followed by a regular reverb trail on another track panned right. The result is a reverb that leads up into the sample and trails away afterwards, while panning across the stage, left to right.
12. Pick out key instruments or sounds and highlight them with reverb while using reverb sparingly, if not at all, on the remaining mix. You may have to adjust reverb send levels as the track progresses so you're not left with the track sounding dry where the reverbed sounds are no longer playing.
13. Usually, bass and reverb don't mix too well, unless you're specifically after a warehouse sound. Unfortunately, this effect results in a loss of definition among the bass regions. Run your reverb returns into a couple of spare channels in your mixer and back off the bass EQ, or add a high-pass plug-in EQ.
14. Don't forget using mono reverbs at times as well. These won't conflict with your rich stereo reverbs.
15. Pre delay is the time taken for the initial reflections to return back from room walls. Use a calculator from www.hitsquad.com/smm to get a pre delay value matched to your tempo. A common technique is to set the predelay to eighth-notes and add the reverb to a straight quarter note kick drum pattern to create an off-beat bouncy feel.