Monitor mix for vocals (with great plugins for free!)

. I will show you my standard vocal monitoring mix with VST effects in Reaper. Among monitor mixes, there’s a very special place for the vocal mix. That’s because singing is a bit different activity than eg. playing drums, keys or guitar. A singer needs to hear them in a proper way which is necessary to keep proper pitch and tone – they particularly sensitive to under- and overestimation or just mistakes.

We’ll set-up a monitor mix in Reaper which will be the basis for the final mix because the elements are similar and after we are done, they’re gonna need only some fine-tuning to sound properly. Let’s start by adding an audio track, turning the monitoring on, choosing the right input and arming this track.

Now we need to construct the effects chain. Those effects are to reinforce the sound, to make it more pleasant to our singer’s ear, to make it fatter, provide fullness, to make it flawless. This chain gets your vocal really wide so the monitoring in headphones is more natural (to prevent pitch and dynamic problems during singing). That’s really important because the better singer sounds in her headphones, the better she sings – so goes the better sounding record!  This composition of VST plugins is quite common for every recording so feel free to make your own preset-chain (or a few). I’ll try to be brief because you could tell literally anything about those effects including writing a book!

What we have hear is chain consisting of great freeware plugins that have been selected by me really carefully, this collection is really a bunch of things you need to have on your PC, serious. I’ve drawn the red arrows that show the signal flow in my monitor mix.

1. Compression – Density MKIII is used to squeeze the sound very firmly. For vocal monitoring I use strong compression with quite low threshold so there’s always more gain reduction than 3dB. Of course it doesn’t stay for the final mix but it usually makes recording easier.
2. BootEQ set to “analog-o-matic” preset that phattens the sound with warm low-mid boost and some lows. Finally it saturates my vocal tracks with some kind of vacuum tube simulation – I don’t care what it is but makes things sound juicier.
3. Ferox – saturation plugin that has to reproduce some magnetic tape units, maybe Studer and stuff. I’ve already used two stages of saturation. I do it purposely – the saturation helps me to deal with transients which may be too aggressive when I use a condenser microphone and all in all, saturating the track helps to bring out more great tone that is hidden in sustain of every note and to make sure I don’t get too loud in the headphones. Just like a compressor but in a rough, more organic, natural way. The two saturation stages let me do it in a less harsh manner.
4. Then we have Classic Delay from Kjaerhus audio. That’s freeware too but the company collapsed, unfortunately, and it’s hard to find a copy (though it’s possible). Delay is set very low, that is only to make monitoring sound fuller. (usually I use here a delay of a 1/8 or 1/4 note, depends on a song)
5. Two times I use Stereo Touch from Voxengo. It’s some kind of artificial stereo but don’t get discouraged – it sounds cool on most of audio material, is totally mono-compatibile and totally nothing wrong about it. I had to grow up to start using it, even though I had a copy on my PC for a few years. In my monitoring chain there are 2 instances of this Stereo Touch, both are very subtle. One is slightly expanding the panorama and the second has longer delay, wider stereo.
6. Finally, a Classic Reverb. It’s from Kjaerhus too. Really good and lightweight suite, those Kjaerhus Classic plugins. I recommend them for anything (monitoring underlined!) because they don’t use much of memory and are extremely responsive! Of course here you can use a SEND reverb as well.

Create perfect performance with fragments of several records. Comping in Reaper.

I will show you how to use Reaper built-in features to perform comping (creating a compile). This time I will use it to refine timing in “doubled” guitar tracks. But it's great technique for vocals, solo guitars and frankly ... everything!

Here’s how my project looks like. There are two tracks of rhythm guitar (red) that will finally work as a stereo group. It’s double to make the guitar part sound fuller and heavier. Both tracks are playing exactly the same part but the trick is that it cannot be done by copy-paste. It has to come from more than one take. Otherwise, some negative effects of comb filtering would appear.

To get perfect double guitar track, we need to perform comping and some other timing treatments.
The tighter your timing is, the better advantage you can get by doubling anything.

• It will sound great
• Nobody will tell there are two guitars
• It will be easier when it comes to mixing!

Before we start comping, I need to record some new takes - right in the place of already existing items. Items are those squares that in other DAWs you’d call regions. Takes are just any records of performance. There can be numerous takes in one item. I just set cursor at 0:00 by hitting home key, make sure if my guitar track is armed and I hit record.

The second guitar track (L GTR) sounds fairly good so there will be no need for heavy surgery – I will just choose some sounds that I like better and then use whole comp as rhythm template for the upper track.

Now we need to decide, how to slice the audio. You can use moments of silence (possibly just rests) to perform clean and safe cuts. It’s good to avoid slicing audio during any sound when it’s not necessary. But I think that the most important is to take phrasing into account, so you don’t lose the natural flow of the original performance.

To slice audio, set cursor anywhere and having the track (and take) selected, press the S button. Before we start, make sure you have auto-crossfade feature enabled. That’s useful for avoiding nasty pops and other side effects of editing audio.

I decided to slice my track regardless of the rests, because the phrases in this riff repeat every bar. Now I listen to both versions and compare. You can switch between the takes by switching the T button. (you can just click on a take to set it active)



Now you can glue the comps so they will be stored in a single file. Use the right mouse button and drag selection over all the items on L GTR track and then press G (glue). We are done with this one and now we will edit the second track so they both fit perfect together. This will be a bit more difficult. Let’s get those three takes nice and large. Save some space for L GTR track so we can use it as reference. Buttons marked with yellow arrow are useful to set the screen look. Mouse roller is great for zooming the horizontal line quickly.

Almost every timing shift that you can see on the screen is audible to some extent. Even if not audible in a rhythmical way, a mass of minor timing shifts can get your mix sound muddy.

Let’s disable the snap-to-grid feature by clicking one of buttons in the left-hand upper corner. This is to make more precise slicing possible as the rest are often not on bar lines.

This sound on the left, was perfectly finished during every take but only the first one was attacked on time. So I slice the guitar part on the rests and change active take … just for this one chord!



Apart from good hearing, you can make a great use of your visual perception of where exactly are similar moments in different takes. You can develop this skill just by working with comps and slicing. Let’s get back to the guitar. Here comes the trouble. Below, in this moment marked with time selection, none of tracks has good timing.

We’ll need to fit one of the takes to work with the lower track (L GTR). We will need to slice this take in-sound. I figured out that there are two chords (look on the left). Anyway, the lowest take (cyan) looks like an unstable chord. The halves of wave shape aren’t even similar and this makes any changes harder to hide (we’ll be probably better off if we slice a sound of rather constant amplitude, tone and timbre)



I took a look at the first take of the upper track (R GTR). I slice (S) this item at three points: during the rest before the first chord, at the point right before the attack of the second chord and during the rest after release. Now focus on your reference track (L GTR) and set markers for attack of 1st chord, attack of 2nd chord and release. To make a marker, set your cursor anywhere and press M. We need those lines to set audio on a perfect time point.

It should finally look like here on the right.

Now we will shift the takes inside items so they appear in the speakers just on time. Note that this affects only a take, not the whole item. That’s really good because it doesn’t mess up whole item and you can still switch between takes.

Select the first take, press ALT key and hold it down. Click on the waveshape and move it horizontally set attack of the first chord to the marker. It turns out that release fits perfect too. If it didn’t, another slicing would be necessary.

As a side effect of this treatment, some sound that was next (or earlier) may appear in your item. Here, the attack of the second chord is about to be played twice - we need to fix that. Just click between the items and move the slice so it’t between two “constant” waveshapes.

BEFORE:

AFTER:

Working with takes and comps in Reaper is mostly just about getting used to this procedure and finding ways to make it quicker, to hear more and to see more on the waveforms. There are a couple of other interesting features related to comping that would improve your workflow.

I hope that you found my lesson useful because that’s the real reason for making it. Let me know what you think. Your feedback will be appreciated! Have a nice day.

Recording electric guitar and bass direct. Impulse response technique + others.

Direct recording is recording a raw guitar signal that wasn't recorded through a microphone. It's done either by plugging a guitar cord straight into an interface or running the guitar signal through some electric unit such as preamp or dibox. I will explain exact direct techniques below.

The equipment needed to record guitars direct is:

• guitar
• quarter inch TS jack cables
• computer with any DAW software (e.g. Reaper) 
• audio interface
• optional equipment

• some additional ¼” TS jack cables

• good quality Direct Box (DI)

• guitar amplifier (combo or stack)

Turning input gain knob on the interface to the minimum before plugging any connector into the signal path or before removing units is a good habit to avoid nasty peaks that would damage your hearing or equipment. Sometimes, even turning down the monitors’ volume too would be a good habit)

Electric guitar is usually being connected using a ¼” TS jack cable. What is common for all the techniques below is that always before recording we’ll need to match the input gain level with a knob on the interface so the loudest part in a song won’t reach -0dBFS or even go anywhere near “the red”. It applies both to LED indicators or displays on interface’s enclosure and meters on the computer’s screen.


Version 1: Connect an electric guitar or a bass with DiBox for more responsive, zero-latency monitoring by a combo guitar amplifier.
To do this, you need a DiBox, additional two ¼” TS jack cables, and of course - the amp. Start out launching the DAW and connect the guitar to the input of a DiBox. The first output of the DiBox should be connected to input of the interface and the second goes to the amplifier’s guitar input. Set the levels as explained above, then turn on the amp, set it’s volume and record guitar parts using no software monitoring “in the box” that would cause latency.

Version 2: Use your amplifier’s preamp tone in the recording - apart from the amp, one additional quarter inch TS jack cable is necessary. This simple technique is about connecting guitar amplifier’s line output to the interface. Always check twice if you chose the right socket and never connect any line-level unit to the speaker output which has very high level that will most likely damage any processor or interface connected to it. This signal is at the line level, so it demands line input in the interface - it’s not designed to be ran through any guitar input. Bear in mind that guitar speaker has a huge impact on how your amp sounds and the only thing you can get from the line output is speaker emulation circuit that in my opinion doesn’t help much to get professional-sounding guitar parts. Although, you would try recording guitar parts that way and then, processing them using Impulse Response technique that I will explain below.

Version 3: Connecting an electric guitar or bass straight into interface to record parts using VST plugins and IRs for monitoring.



The first step is launching the DAW, connecting your guitar to the guitar input in your interface and setting suitable level. Then we’ll need to build FX chain to simulate guitar amplifier and speaker cabinet. To do this, I will use:

• · Le456 – a freeware VST plugin that’s a good simulation of a guitar amplifier. (http://lepouplugins.blogspot.com)
• · LeCab 2 – a freeware VST plugin that’s capable of reproducing a timbre of any speaker cabinet using specially prepared, short Wave files (called Impulse Responses) (http://lepouplugins.blogspot.com)
• · RedWirez impulse response samples are available as a free demo pack that consists of some really nice Celestion-packed Marshall 1960 tones. (More IRs in the paid version) (http://www.redwirez.com/free1960g12m25s.jsp?ref=home)

The rules of building an FX chain for virtual guitar amp are the same as for the real one. Choose a track in your DAW and add first VST plugin: Le456 and next: LeCab. Now we need to configure LeCab to work as a cabinet simulation. The LeCab 2 consists of 6 sections that remind of channel strips. You can use only one to make it work but two or more are necessary to get a decent guitar tone.

What are those controls for? The button with a folder icon is for opening an impulse response file (that you downloaded from RedWirez website). You will sort out quickly what all those names are for but the key is microphone placement. It works on the same basis as picking up electric guitar through a guitar amp with a microphone in front of it. The LEFT/RIGHT input switch on the left is for choosing from which stereo channel will the raw guitar be taken from (in case you would like to record two guitar parts and build a stereo group of them). All the other controls are for how the impulse responses are used in the finally processed sound.

The best way to get a good tone this way is to fine-tune your Le456 and LeCab settings tweaking controls during playing guitar (or listening to playback of a track recorded before). So go to your DAW track and arm it for recording so the track input works (I marked it with orange square) and enable monitoring so you can hear what you play in monitors (yellow square).

What we record is just a raw guitar track – as it comes to the input of the interface, the DAW is applying the simulations of guitar ramp tone. It means that you can improve tone of your guitar anytime during mixing, such ways as changing amp to another VST plugin, switching between impulses or even reamping (which is rather a long story to tell).

We took some closer look on the most popular direct recording techniques. Each one has some pros and cons and It’s really worth it to know all of them to pick up the suitable method for any studio situation. While recording through a DiBox and monitoring in a combo amp seems to be the most reliable and very elastic solution, the third of techniques described in this lesson, using software monitoring and Impulse Responses may be applied at really low latencies that wouldn’t make a difference and it doesn’t need that many pieces of equipment while giving similar results. Optimizing the OS of your computer can really make a difference. But remember that it depends both on the equipment and it’s configuration so it’s worth to spend some time on it, to get good results. In my opinion, recording parts through guitar amp’s line out may be really great for bass guitar, but electric guitar takes more effort due to the necessity to add some processing that would reproduce filtering that normally comes from speaker. In my opinion, using amplifiers line-out is rather a bad idea, unless you have a really great preamp that’s worth it – then Impulse Responses would come into play, maybe bringing some stunning results…

Studio monitors. Do I need them? What can I use instead?

Obviously, people who can afford, should go and buy monitors. They will be able to reproduce audio so precisely that you can be very comfortable about making changes to audio and being sure about it's translation to different audio systems (translation is quality of how exact is audio reprodution on different audio systems) But if you are on budget, or you still don't have all the necessary microphones to get stuff done, it would be better to settle for some consumer audio.

The most crucial quality of monitors is perfect crossover between LF speaker and HF speaker. (a circuit responsible for sending lower frequencies to one speaker and higher to the second). The typical problem about that is how precisely is this circuit calculated. This calculation involves data that can be obtained only by measuring actual frequency response (as air pressure).

So if the big speaker is playing everything below 600hz and the little speaker everything above 600hz, the total frequency response of your monitors should be totally flat at 600hz. (no amplification and no attenuation!) Thats difficult because obviously responses of those speakers will overlap. Most of consumer audios would have a serious bump here.

Here's a picture that shows how can a peak appear on the crossover frequency.

Audio crossover graph. Please don't care about exact numbers, just take a brief look to catch the idea.

Most of cases, frequency response of a consumer set of 2.0 speakers will be not totally flat but resonably predictable As soon as you pick up a pair of speakers that don't have any kind of artificial boosting, exciters etc. You just need to listen to a lot of music with those speakers you are working with. Remember that folks who use monitors, need to "learn" their monitors too!

This kind of speakers (2.0) doesn't have much bass and very high end. That's right.

But listening to everything that those little speakers can deliver, is essential to a good sounding mix.

You need to take care if you manage your bass right and it's not that hard because if you ever get too much bass, those speakers will lack energy etc. It's also useful to check it with headphones sometimes.

Here are some audio checks that I recommend you to do before any public release of your tunes. Those are very good things regardless of what monitoring system you are using (even if you have monitors, it's good to give your song a listen with eg. a little speaker of smartphone)

Some additional checks to make sure your mix has fine "energy management" 

• play it on a car audio 
• play it on a home cinema with a huge subwoofer 

Make sure that volumes of your instruments are perfect and there are no bad details:

• download the song to your smartphone and listen with the built in speaker
• listen to the song on built-in notebook speakers

Check for excessive hi-end:

• listen to it on an iPod with sound-isolating earphones (those that go deep into ear)

Recently I have read an article in Sound on Sound, that was about a famous producer who mixes on a little hi-fi ! In spite of real, good monitors waiting in front of him. At the moment I use a cheap "hi-fi" too. I am about to release an album that will be probably mixed with those speakers. I'm moving soon and don't wanna buy any big toys.

Get what you have (the most natural sounding audio from those you have, preferably 2.0) and go make things!