Recording Starting out with Reverb & Compression

Reverb​

Why reverb?​

There are two main reasons you might add reverb:

• As an effect to make the music sound "subjectively" better. In this case you, or the audience, hear the reverb as something that can add more life or sparkle to the sound. It is easy to overdo this effect and swamp the recording. This is often the case with beginners, but also we old pros can sometimes be guilty, especially if we just got a new reverb unit or plugin and go a bit overboard.
• As a way to make a dry recording sound more realistic. If your recording room is very dry, or the instruments/voice are recorded very close, it can sound a bit odd because our ears are used to hearing the ambience of a room.
• A combination of the two.

Setting up on busses​

First of all this assumes that you want to have the same type of reverb on all your intruments. If not, then you might as well just use separate reverb plugin on each channel’s inserts.

However very often to get a mix hanging together nicely then you do want the same reverb or combination on all or many of the instruments in your mix.

After all, the point of adding reverb often is that you are making an instrument or voice sound like it is in a room with its natural ambience. So it makes sense they are all in the same room.

What I do is put one reverb on a separate “bus” in Logic Pro. This is not the same as a channel, because a channel can only have MIDI or audio as an input source. A bus is similar, but its input signal can be any audio “routed” from the sends of any of your channels.

The reverb on this bus is set to 100% wet, so it’s only reverb (no dry signal).

The outputs of all the channels are set to go to your main mix.

The output of the reverb bus is also set to the main mix, so the dry signals from all the channels meet up with the reverb output of the bus and get mixed together.

As the sends from the channel strips send a dry signal to the reverb it’s very easy to send different amounts from different channels to the same reverb. You may want a very slight touch of reverb on a piano, but a bit more on a vocal, so you just send more to the vocal via the bus send. And when you change the reverb type on the bus, it applies to every instrument that is sending there.

Reverb on buses

In the example above the Audio and instrument track each have a rotary send output in addition to the main outputs to the stereo mix. These go to the room and plate reverbs on the busses 1 & 2 which in turn send the ("wet") signal via their main outputs to add to the stereo mix.

The way I work I often have two reverbs as I mentioned earlier, and so I have two busses and two sends on each channel. Bus 1 is a short room (about 0.9 with a slight predelay) and bus 2 is the longer hall or plate reverb. Send 1 on each channel goes to bus 1 and I usually have all of them sending at least a little bit. This is especially useful when using MIDI samples to combine with an instrument or vocal recorded in a room, as you can get the very dry MIDI sounding more like it’s “in the room” with the real instruments.

Then I add the bigger reverb as desired. Again the send control is all I need to determine how much, no need to open the plugin and dick with the settings.

Logic template - 2 reverb busses​

Reverb tips:​

• It can often work well to add a short (0.7 – 1.2 sec) room ambience to dry electronic signals (e.g. synth modules, D.l.’d guitar) .This simulates (to a limited extent) the effect of recording in a room. You can then add a longer reverb (e.g. 2 secs and upwards) as necessary for effect. This also works on close miked acoustic signals. This method gives your mix a lot of life without using too much reverb (a fault of many “home demos”). My autoload file has a short room set up on bus return 1 and a longer reverb on return 2.
• If you want reverb without making a voice or instrument sound to distant, give it some predelay (50-70 milliseconds). This simulates the effect of sitting close to a stage and hearing the signal first then the reverb from the back, as opposed to sitting far away from the stage and hearing the reverb and signal at the same time.
• To simulate distance set the predelay to very short. This simulates sitting at the back of a hall and hearing the sound from the stage almost at the same time as the echo from the back wall. Sounds from further away will often send less bright than close sounds, so it can help to filter or EQ some top off.

Audio Compression​

Basics​

The aim of a compressor in recording is to reduce the range of dynamics of an audio signal. (Don’t confuse this with file compression, which is used to make a computer file smaller). Any parts of the signal louder than a certain THRESHOLD are reduced. The amount of reduction is relative to the level of the signal and expressed as a RATIO. If the ratio is set at 2:1 the signal above the threshold is reduced to half its original, if the ratio is set to 5:1 the signal above the threshold is reduced to one fifth of its original.

(NB if the ratio is set as high as possible, usually infinity:1, the signal above the threshold is reduced to practically the same as the threshold and is referred to as LIMITING.

In this graph a 45^o
line represents equal input and output levels, ie no compression (1:1) ratio.

The amount of compression is based on a relationship between the threshold and the ratio. Low thresholds and high ratios give you more gain reduction, i.e. more compression, but neither threshold nor ratio on their own will determine the amount of gain reduction. You can achieve the same gain reduction by a high ratio and a high threshold as you can with a low ratio and a low threshold:

As you can see from the above, the amount of compression with 7:1 ratio and high threshold can be similar to the amount of compression with a 2:1 ratio and lower threshold, and is actually the same at one point. Whether to use high ratios or low threshold or both is a subjective decision and usually based on trial and error and taste.

This is not the end of the story. Usually the point of compressing a signal is to get the whole signal louder. Most compressors have a gain reduction meter. Sometimes there is a switch to change an input gain meter into a gain reduction (GR) meter. You can tell from this how much compression has been applied. Having reduced the loud bits by a certain amount, you can then adjust the gain make up by the same amount. This effectively brings the loud bits back to their original level, but at the same time brings the quiet bits up to a level higher than they were originally, with the end result of reduced dynamic range and higher overall average level. (Many software compressors can do this automatically)

Why Compress?​

A wide dynamic range is often ideal in a live acoustic performance, however recording and broadcast media generally have limitations. Peaks of audio need to be below the level at which distortion occurs. The grooves of a vinyl record can only cope with so much level until the needle jumps. Digital media only allow levels up to 0dB. Analogue tape causes distortion when the level is too high. With very dynamic music, once the peak level is set at a practical maximum, the low levels may well be close to the level of unwanted noise, e.g. tape hiss, vinyl surface noise, background sounds. Many radio stations compress the output signal as very quiet passages of classical or acoustic music can become totally inaudible on car radios that have to compete with the car’s engine noise and the sounds of other traffic.

• Compressing a final mix will make it louder, and therefore possibly sound better when compared with less compressed tracks, e.g. on radio or on compilation albums.
• It can be useful to compress single tracks of a multitrack recording to avoid quiet words of a vocal, or quiet notes of a solo instrument getting swamped by the backing.
• A combination of compression and automatic gain make-up on instruments with a natural decay will create a sustained sound.

Hard or Soft?​

In the graphs above, compression occurs exactly as the signal hits the threshold. Some compressors gradually apply compression as the signal approaches and after it has crossed the threshold. This is often described as soft knee compression. The graph can be seen as the thigh, knee and calf of a leg:

Speed​

There are often two more controls on a compressor, ATTACK and RELEASE. The speed at which the level decreases as the signal crosses the threshold is determined by the Attack control, and the speed at which the level increases after the signal drops below the threshold is determined by the Release control.

A fast attack means that the full amount of compression kicks in almost immediately. I found this can sometimes cause distortion or have an adverse effect on the tone of an instrument or the amount of punch, especially bass drums. The best settings are found by trial and error, but a good starting point is a mediumish 100ms. With a slower attack, the compression gradually kicks in.

The next two pictures show what happens when a signal is is passed through the digital compressor in Apple Logic Pro, with a threshold of -10dB and a ratio of 2:1. The signal starts just below the threshold at -11dB, then suddenly goes up to 0dB and back again.(A) shows the signal with no compression. (B) shows the same signal when compressed with attack of 200ms and release of 500ms.

(a) No compression:

(b) With compression:

Exampes of different compressions on a saxophone​

View: https://www.youtube.com/watch?v=4arEEumOfZY

Tips & Hints​

• Start with a ratio of between 2:1 and 7:1, medium-fast attack and medium release then gradually lower the threshold until you get gain reduction of about 5 dB. You can then set the output gain make up to compensate, eg if your gain reduction is 5 dB, set the gain make up the same to bring the peak level back to its original. Then gradually speed up attack until it gets noticeable and back it off slightly.
• Listen carefully. Compression can affect the timbre of an instrument, either because of the inherent sound of the compressor, or because the peaks of an instrument may have a different tone to the troughs, so reduction in level of the peaks relative to the troughs changes the overall tone. This can be especially true when applying fast compression to instruments with broad vibrato.
• Be careful when compressing an entire mix. Very often in pop music there is a bass line which is generally constant in level. If there are sudden very loud peaks such as brass stabs, the compressor will lower the whole track at that point, with the result that although you may get an overall levelling of dynamics and increased loudness, the bass line will dip at this point and lose its flow. This is sometimes called pumping. This problem can be overcome by using a multiband compressor, which splits the signal into different frequency ranges, and compresses them individually. In the above case the brass stab would be compressed, but the lower frequencies can pass through a lower band of the compressor with no or very little compression. It is usually best to mix a track with no, or very gentle, compression, then apply compression at a later stage (final mastering).
• If you are using compression on a vocalist, ask them not to back off the microphone for loud bits. Singers often do this when singing live, but usually a constant distance from the microphone is better in a studio. If they suddenly back off, the vocal sound will suddenly get more ambient which may not be good if you want an “in your face” sound. A good compressor will cope with a large range of dynamics without changing the sound drastically.

 

[Ballymenaboy]

Brilliant article and most helpful
I use a ZoomR20 for recording sax over backing tracks and add the effects such as reverb post recording,then mix down and export to laptop.
Not being very IT literate a multi track recorder is easier for me

 

[LostCircuits]

Got it, it's a good start on how the basics work, I had hoped to find a little bit more of a cooking recipe, potentially with some short sound bytes.

And I am talking more about some "quick start" guide, like for example you take a dry recording and apply compression (attack-release, input-output and mix which are sort of obvious). And then there are the more critical settings like low/band/hi pass to eliminate or suppress some frequencies.

Add EQ to add warmth and suppress the nasal components in the 2500 Hz range.

The main reason I am bringing this up is because you have a great sound on the sax. I don't care about vocals right now, and midi stuff, I can handle but I am struggling with getting, for example, tenor recordings to sound the way I hear it when I record the track.

I know @mrblackbat has a really good handle on these things, too, so what I am trying to get at is a saxophone-specific "mastering" thread but it seems a bit like secret sauce 😛

 

[Pete Thomas]

And then there are the more critical settings like low/band/hi pass to eliminate or suppress some frequencies.

If you are still talking about compression, as opposed to just filtering, then that is multi-band compression, a more complex thing altogether. It is a good way of EQing but is more complex than either "normal" single band compression or parametric EQ.

I do use a multi band compressor (sometimes with 5 separate bands) but generally as part of a final mix or mastering.

There is no saxophone specific effect although I do use filters (low and or high pass) as opposed to immediately reaching for a parametric EQ because for me it's the best way to give each instrument a space in a mix spectrum.

For example if there is a solo saxophone, I would hope to leave it un-EQed if the recording went well. But wherever it overlaps other instruments I might use a filter. If the saxophone has some bass frequencies overlapping with a bass guitar, I might apply a tiny bit of gentle high pass to de-muddy the mix at that point.

I found this to be more a thing with congas. If there is a conga solo you want that low conga to boom a bit, but in a track it can completely muddy the bottom end without actually adding anything musical, hence high pass filter it a bit.

I've learned a lot of this over the years from doing mixes, then sending them off for mastering and getting feedback and advice from great mastering engineers. I would often attend the mastering sessions to learn where I could improve. As I got better at mixing, the less they had to "fix" in the mastering studio or dubbing suite.

 

[LostCircuits]

Thanks Pete, this is getting to the bottom of it:

I've learned a lot of this over the years from doing mixes, then sending them off for mastering and getting feedback and advice from great mastering engineers. I would often attend the mastering sessions to learn where I could improve. As I got better at mixing, the less they had to "fix" in the mastering studio or dubbing suite.

I know, one pinch of salt may not suffice and two will ruin the stew, so it really comes down to experimenting and being careful about what you apply and how to record and mix and at what stage you apply what. I wish there were some local courses offered here - lots of musicians and even some studios but everybody is playing very close to their vest.

But I already learned a few things just reading this thread and I hope we can keep it going with more input from others.

 

[Pete Thomas]

potentially with some short sound bytes.

The thing is if you use compression the way I would recommend, you don’t hear anything different. It just makes all the notes not get stick out or get lost in a mix.

It evens out the dynamics without losing any of the expression associated with dynamics.

 

[mrblackbat]

but I am struggling with getting, for example, tenor recordings to sound the way I hear it when I record the

It all starts with good tracking. A good microphone, positioned well, in a good sounding room (either one that has pleasing acoustics or is so dry that you have full option to do what you like later) is going to help massively.

Then it's about trying to place the sound into a concept of space. How do you want the listener to perceive the saxophone? As if they are right up next to it, or if they're a bit further away? If there are other instruments, how does the sax relate to those, and you can carve out space for each one - both in terms of frequency to avoid muddying and clashing, and in terms of physical space, with panning and reverb (and also eq again here - the further away something sounds the less top end it has etc).

Compression is it's own beast, and can also be used as eq - for instance if you have a slow attack then the resulting sound will be harder and more focused on the transient, a fast attack softens off that transient. Release then controls how natural that will sound; a common mistake I find for saxophone recordings is a very fast attack and a very fast release which results in quite a nasally sound.

 

[LostCircuits]

Release then controls how natural that will sound; a common mistake I find for saxophone recordings is a very fast attack and a very fast release which results in quite a nasally sound.

Thanks Matt, this is exactly the kind of stuff I was trying to steer this thread to. The funny thing is I have no problem with soprano (after endless hours of playing with the mic positioning etc.). And of course, there is the operator issue, you put crap in, you get crap out no matter how you try to make it shine.


Edit: The attack and release were the culprits. It's like the curtains were drawn open. Now I can concentrate on playing, at least start again.

 

[mrblackbat]

So play with those settings and learn to hear what they doing.

Have a very fast attack - then adjust release from very fast to very slow and listen to how it shapes the sound. Then do the same for slow attack.

There's a great video by Gregory Scott on learning to hear compression; it's on drums but will give you a good idea of what's happening:

View: https://youtu.be/K0XGXz6SHco?si=a6QCpNS3X43x128K

 

[Pete Thomas]

For me, compression on a saxophone became more or less redundant for mixing once I was able to use automated fader control [EDIT: I later found it was better to automate a gain shaper or gain] plugin. This is because you get absolute control over the level over time. It takes a bit longer as it isn’t just a setting on threshold and ratio (that controls the smoothing out of dynamics) along with any inherent problems specific to certain instruments with a pronounced attack.

However if you wanted to accentuate the articulation, slow compressor attack will do that. Traditionally a fast attack will compress the articulation along with the rest of the signal but obviously had to wait for the signal in order to do it, hence even with fast attack a teeny bit gets through first and with higher ratios can have an unwanted effect.

Basically slow attack compression on a saxophone can really accentuate the tonguing.

Wouldn’t it be good if compressors could predict the signal and so operate in advance and negate this issue?

Well these days they can, via “look ahead” and this is a feature of some plugins. Obviously it can only work on previously recorded material because it needs to see what it is.

But even so I still definitely prefer using fader/gain manipulation along with automation for better control. I could say, for me, there is no longer any need to use a compressor on saxophone, vocals etc unless for a specific effect.

 

[Pete Effamy]

automated fader control

The old-fashioned “riding the faders” , done in real time as the session was being recorded. Sinatra’s vocals were recorded like this, with the engineer looking out for Frank belting something out.

Which automated signal rider do you use Pete? I have the Waves one, and I’m convinced that it changes the sound a little.

 

[Pete Effamy]

As has been said, compression becomes a necessity really when there is a busy mix, helping to push a solo instrument through rather than having to turn it up too high. Gain staging (taking original signal down in order to create headroom for effects to be applied - which add gain) is a must.
Big solos / vocals with few or no dynamic range will likely be treated differently to say a jazz quartet, which might be full of dynamics and nuance. Not much to fight through though so effects can be whittled down considerably.

 

[Pete Thomas]

Which automated signal rider do you use Pete?

I now just use the track automation on the gain plugin in Logic. Generally an automation curve that is the opposite of the wave form levels, then tweak it a bit to taste within the mix.

There used to be a dedicated plugin called gainshaper I think, it did the opposite so you could draw an automation curve that mirrored the waveform.

Look ahead compression (in theory) would do the same provided no attack or release, however I like the tweak ability of an automation curve.

 

[Pete Thomas]

Here is an example of automating the Gain Shaper plugin on a saxophone track. The first part is quite extreme and brings sup the quieter parts(for the first section) in a way I preferred, I could not achieve this with a compressor

Doing the same with the native Logic gain plugin would look like that but upside down.

Here's a track where I just used the gain, less of an extreme treatment that brings up just some of the note "tails":

 

[Pete Effamy]

Here is an example of automating the Gain Shaper plugin on a saxophone track. The first part is quite extreme and brings sup the quieter parts(for the first section) in a way I preferred, I could not achieve this with a compressorView attachment 29270
Doing the same with the native Logic gain plugin would look like that but upside down.

Here's a track where I just used the gain, less of an extreme treatment that brings up just some of the note "tails":

View attachment 29271

Fab. I have no knowledge of it - will look into it.

 

[Pete Thomas]

Fab. I have no knowledge of it - will look into it.

I don’t think you can get it any more, hence I use the Logic gain plugin instead.

 

[mrblackbat]

For me, compression on a saxophone became more or less redundant for mixing once I was able to use automated fader control. This is because you get absolute control over the level over time. It takes a bit longer as it isn’t just a setting on threshold and ratio (that controls the smoothing out of dynamics) along with any inherent problems specific to certain instruments with a pronounced attack.

However if you wanted to accentuate the articulation, slow compressor attack will do that. Traditionally a fast attack will compress the articulation along with the rest of the signal but obviously had to wait for the signal in order to do it, hence even with fast attack a teeny bit gets through first and with higher ratios can have an unwanted effect.

Basically slow attack compression on a saxophone can really accentuate the tonguing.

Wouldn’t it be good if compressors could predict the signal and so operate in advance and negate this issue?

Well these days they can, via “look ahead” and this is a feature of some plugins. Obviously it can only work on previously recorded material because it needs to see what it is.

But even so I still definitely prefer using fader manipulation along with automation for better control. I could say, for me, there is no longer any need to use a compressor on saxophone, vocals etc unless for a specific effect.

I definitely agree that if you're just trying to control level, then fader control is all you need.

Compression for me is there to shape the sound, and often adding the colour of the compressor itself to the signal, especially when used in parallel processing.

 

[Pete Effamy]

I don’t think you can get it any more, hence I use the Logic gain plugin instead.

I know it - it can be assigned automation then?

 

[Pete Effamy]

I know it - it can be assigned automation then?

Gain plugin or automated fader? Gain plugin is effectively what I do with gain staging - not always a carte Blanche whole track thing - especially vocals

 

[Pete Thomas]

Yes, any parameter of any plugin can be assigned automation

I know I said fader automation earlier, which I used to use at first, however it's not so good as using the gain plugin, which does the same job as it affects the output of the track, but it saves having the downsides of fader automation because with the gain plugin, you can still manually use the fader to adjust the overall volume of one track relative to the other tracks.

If you automate the fader, you are not able to do that easily.