Composer Focus - Music & Audio Tutorials

Pimp my Drums - Kick & Snare

Mon, 28 Nov 2011 19:42:03 +0000

You definitely will need a good and solid drum performance (and also a good and clear recording signal) to get good overall results afterwards. But a good recording does not equal good sound necessarily. The first important thing is that you record your drums on separated audio tracks. Make sure that there are individual tracks for kick drum, snare drum, hihat, overheads and ambience (if you want to catch the sound of the recording room as well).

Listen to sample1. These are the naked drum tracks with no editing at all.

Then, please switch to sample2. This is the result I got after doing the steps I explain below.

The Kick Drum

The kick drum is the basis of every groove and therefore very important. Make sure that it has its own place in the mix without competing with other instruments (e.g. the bass guitar). I would like to show you my chain of insert effects to let you know what I've done.

1) UAD Studer A800 (Tape recorder)

The Studer is a great tool to get an analog vibe. It is designed to be put on every single audio channel in your song. I picked the preset for kick drum and adjusted the input and output levels. In order to run the Studer on your own system you will need a UAD-2 DSP card. Obviously, it's a matter of taste and whether or not you like the sound of a tape recorder but I use it on almost every song.

The Studer usually should be used in the very first slot of your effect chain because its tape saturation adds some color to the sound.

2) UAD 1176LN (Compressor)

To give some stability to the kick drum we need to compress it. I could have used any other compressor but I like the 1176 because it offers very fast attack and release times that give you the possibility to shape the sound of the kick.

Furthermore, the 1176 allows you to really overdo the compression. This will make the sound become dirty and a bit distorted – this can be a useful effect if you were looking for a lo-fi drum sound.

3) Studio-EQ (Equalizer)

Using the equalizer I support the low pitch of the kick to give it a bit more low-end and also more sharpness by increasing the frequencies at around 7.6kHz.

Every kick drum has its own frequency points that are important for the sound characteristics. But in general there are two important spots to look at: a) the low pitch (see band2) and /cool.gif' class='bbc_emoticon' alt='B)' /> the snap (see band4). Find those spots and you are ready to shape the
sound properly.

4) Gate

The only purpose of the gate is to get rid of noises and sounds other than the kick drum. This is important because I am going to use the signal of the kick drum to trigger another signal afterwards (see No.6). I suggest finding the quietest hits of the kick and adjusting the threshold to make sure those hits get through the gate.

As you can see, I started with a preset. But I highly suggest experimenting with the settings because especially the attack time and the release time have much influence on the sound of the kick.

5) UAD Precision Enhancer Hz (Special processing)

This tool provides different modes for different sound sources. Mode D is designed especially for kick drums. It creates the illusion of more and a deeper bass due to the processing of harmonics. This is often associated with the term “phantom bass”.

Always trust your ears and also keep an eye on the volume meter when using bass enhancers in general. On the one hand, it can be pure magic but on the other hand they certainly have enough potential and power to mess up your mix totally. Therefore, my beloved knob on this tool is “effect” to find the right amount of phantom bass and original bass. That is the best way to find a good balance between both signals. Again, it can be a great effect (and also sound) if you just used the phantom bass signal in your
mix.

6) Sub bass (via side-chain)

This last step probably has the biggest effect on the sound. I recorded a simple sine tone and put it on a separate audio track (listen to sample3). I used a frequency that cut through the song (unfortunately I can't tell anymore what frequency that was). Next, I used a gate in side-chain mode on the sub bass audio track.

The original kick signal feeds into that gate and functions as a trigger. Every time the kick hits the gate will open and the sine tone is audible until the release time closes the gate.

The Snare Drum

In my opinion, the snare drum is equally important to the kick although there are some different schools of thought. Some say the snare drum is the most important thing in a good mix and therefore it has to be the loudest source.

1) UAD Studer A800 (Tape Recorder)

As mentioned before, I usually put the Studer on every audio channel and the snare drum is no exception. Since you will not find a preset for the snare drum, I used the settings for “saturation” to make sure it has enough punch and presence in the mix. Again, this will change the tonal color of the original recording so make sure to use the Studer in the first slot of your effects chain.

To give the best emulation of the hardware device, Universal Audio even included the noises of the machine. This is good for realism but sometimes you would want to get rid of them. Just click on “open” and change the settings according to your needs.

2) Gate

As we did with the kick drum, we will separate the snare drum signal and get rid of crosstalks. Since we use the signal to trigger some other sounds it is important that you make sure, every snare drum hit breaks through the gate. This won't be as easy as with
the kick drum because the snare usually is much more dynamic but this also depends on the musical style and the drummer.

Most gates allow you to define a center frequency that opens the gate.

3) Studio-EQ (Equalizer)

Next, I have shaped the sound using an EQ. The presets are a good starting point. You might want to adjust some setting depending on the sound of your snare.

I would like to point out the most important band is the low cut. You might want to leave most of the low end to the kick drum and the electric bass. This prevents your mix from sounding muddy.

4) White noise (via side-chain)

Now, the snare drum sounds are already satisfactory, but still not expensive. Therefore, I am going to add another audio track to trigger a white noise signal with the snare hits. Listen to sample4, this is the signal that I am going to layer on top of the snare drum. I am using the same signal flow like on the sub bass channel for supporting the kick drum.

5) Some fancy noise (via side-chain)

To let the snare drum appear a bit more aggressive, I am adding another noise to the hits. Listen to sample5. If you own a copy of Spectrasonic's Omnisphere you are going to find many extremely cool sounds that serve that purpose (not only that purpose, it really is one of my go-to plugins for almost everything.

Here you have an overview of all active drum tracks. The first two from above are the kick drum, followed by three tracks for the snare drum. The rest is a separate track for the hihat and two overheads (left and right). The last track was intended to catch the sound of the room and therefore it adds some depth and ambience. There are literally endless possibilities to manipulate and edit the sounds of each track. The only important thing is that you find a sound setting you like best. Most of the time, you will find those settings by experimenting.

Reverb

The last step is adding some reverbs. I set up an effect channel (send-return) and loaded the Altiverb6 from AudioEase on that channel.

I am not giving too much reverb to the kick drum because I want to leave it in the foreground of the song. The snare drum gets more reverb. It is important to keep an eye of the tempo. The slower a track the more reverb you can put on the instruments. This is
a general rule of thumb. You can also experiment with different reverbs for different tracks. This way you can create more depths in the drums that can make them more transparent at times. It is all about experimenting.

How You Can Achieve A More Expensive Sound

Fri, 08 Jul 2011 17:38:59 +0000

What does ‘expensive’ mean?

Although there are some limited ways to correct the sound of a mix down in the mastering process, this tutorial is not about mastering nor mixing. It’s about getting single tracks to sound great by themselves in the first place to avoid spending a lot of time correcting mixing mistakes later. Actually, this is the first step even before the mixing process starts. Make sure that you have access to all the single tracks in your song since it will be necessary to edit every instrument differently.

In order to show what our goal is, in a practical manner, we first need to define the word ‘expensive’ in a musical context. Listen to example1 (preferably on good headphones since we would like to get a very close look at the frequencies) with the volume reduced to a minimum.

I switch between two different versions of the same song every two bars; at low volume the difference between both versions should not be so big.
You should be able to tell what instruments are being used and what they are playing no matter what version you listen to.
So why should I move on with this tutorial anyway?

Important: If you want people to listen to your music, and especially if you want to have success with your music, the above described case isn’t reality. Usually people listen to music loudly because their brain tends to equate loudness with quality (compare this with cinemas, theaters, concerts or even car audio systems). If something doesn’t sound good on high volume then there is no quality involved.

Listen to example1 again (on the same headphones); however, now turn the volume up but be careful of your ears. Now, some frequencies become very disturbing; they even hurt. So this shows that our task is to make the music sound great at high volume, not at low volume. Although one versions sounds brighter, this is not achieved by simply increasing the high frequencies. It is a bit more complicated than that, as we will see.

It is advisable to listen to some modern pop or rock productions, since they are a superior example of quality in the above described context. Additionally, if something sounds excellent at high volume then it usually will sound excellent on low volume, also.

Manipulating Frequencies

Without a doubt, getting a rich and expensive sound is the result of many decisions that must be correctly made during the entire production process, because every correct decision adds a bit more quality to the track. However, let’s focus on just one option for now.

Note: Since our ears do not work linearly, it is to be expected that some frequencies will suddenly become more pronounced when the volume goes up. However, I don’t want to get too deeply into physics because we would like to keep it practical at this point.

Getting the right sound is always about manipulating frequencies. There are many tools at our disposal to achieve this, such as: equalizers (EQ), enhancers, loudness maximizers and many more … even reverberators. I’m going to use the equalizer because:

it comes with every audio editing program
when using the highest Q-factor you may manipulate just a few frequencies
it only manipulates the frequencies and does nothing else but that

The following illustration shows the usual, unedited frequency spectrum of a single instrument on the left and the spectrum we want to create using our EQ on the right.

Obviously, our goal is to flatten the frequency curve of every instrument in order to keep all the frequencies under control at high volume. This way the mix down of several different instruments is much easier because you don’t need to deal with single frequency peaks anymore.

Theory Turned Into Action

The first thing we need to do is find the frequencies that we need to edit. So, activate a band of your EQ and start sweeping through the audio material slowly. You are going to find fundamental frequencies as well as harmonics. Keep in mind that these frequencies are not static… they change according to the tone pitch; the dynamic involved in the playing performance and the articulation; so it might be necessary to automate the EQ settings along the time-line in order to catch the changing frequencies. Of course, this is not true for non-tonal instruments, like percussion and drums.

Once you have found the resonant frequencies, reduce them carefully using the highest Q-factor. Trust your ear and don’t overdo the reduction otherwise the instrument is going to sound very thin and tinny. I assume that the reduction is going to take place somewhere between -3dB and -10dB. It might be helpful to use a real-time spectrum analyzer, like the “Pinguin audio meter” or the “T-Racks3” spectrum analyzer but these tools can’t replace the best device you have at your disposal, your ears.

It could be possible that you will find many frequencies that you want to reduce and one 4-band EQ just can’t catch them all. In that situation, don’t worry about using as many EQ instances as you need to, to achieve the best sounding result. Most sequencers offer a limited amount of insert-fx-slots (usually eight). In order to free those slots after the frequency editing process, record the edited instrument again to a new audio track with all EQs activated. Usually the sequencer software doesn’t allow routing one audio track to another one, so you will have to find a workaround to fix this routing issue. The following picture is about how to solve this problem in Cubase (since version 5, before that free routing was not possible).

Although the CPU isn’t the performance bottleneck that it used to be, we always recommend deactivating all plug-ins on the original instrument track. We won’t need the original track for the playback anymore since it has been re-recorded. Remember, do not give away any CPU power (or even DSP card power) without a good reason.

Note: If your EQ is not a VST3-plugin I highly recommend to deactivate it rather than bypass it. Because when on bypass the CPU still calculates the effect in the background even though it’s not audible. Only deactivating the effect really frees the CPU power again.
Explanation of the example

The example uses the whole described process in an A/B-comparison. One part uses no EQ at all, so it is obvious that there are many frequencies that make the overall sound muddy and nontransparent. The other part uses many instances of EQs on every track. All the frequencies were located by sweeping through the spectrum. The result is that every instrument now has more clarity and doesn’t compete with the other instruments anymore.

As always, please keep in mind that there is no such thing as a Holy Grail in audio production. Excellent audio production is achieved only by experience and a big amount of experimentation. At least, that is what I believe.

Make Your Drums Sound More Punchy

Mon, 06 Jun 2011 10:53:46 +0000

Normally, you would want to use a compressor on your drum track to catch the very loud peaks. But be careful in doing so because if you go too far with this effect you are only going to lose a lot of your dynamic range. Consequently, the sound might become inappropriate for your production. So I am going to share with you a little trick that retains the dynamic range but adds more punch to the sound. This applies to recordings of real drums and virtual instruments.

2 basic ways to affect/edit your drum sound

Basically there are two ways to edit an audio track in general. The first one uses insert effects while the second one uses send effects. Both ways bring their specific advantages into play. Insert effects affect the whole signal that runs through the audio track. On the other hand, send effects affect a copy of the audio material without touching the original signal. That means, the affected audio data is added or returned to the original audio data.

One big advantage of send effects is that you are able to send different signals from various audio tracks to the bus without increasing the CPU workload. But nowadays the CPU is not the bottleneck anymore, neither is RAM. Clearly, it becomes a matter of personal choice and what fits your workflow best.

Note: You can use a send bus as a substitute by duplicating an audio track and using insert effects on one of these tracks only. With the right portion of volume, you can set up a balance between both tracks.

I prefer using send effects in most situations because you can glue your instruments together by using typical effects on send busses like delay or reverb. In this way, you have to set up the effect only once and theoretically you can use it on every audio track in your project.

Adding brightness and punch

In setting up two separate send busses, I can add some brightness or give more punch to the drums. The architecture of send busses are (usually) identical throughout different DAWs no matter which DAW you are using.

Note: The drums I use in my sample are already processed using EQs, compressors and gates. I usually do the mixing process of the drum set first to create a good and solid basis to add further spicy effects. So I (personally) do not consider the send busses being a part of the ordinary mixing process. But (as always) that is a personal point of view.

Brightness: All effects listed and described in this paragraph are used as an insert effect on the first send bus.

First, since we want to edit the high frequencies only we need to set up a HPF (high pass filter) at around 3.1k Hz.
The next effect in the chain is a short ambience reverb (reverb time: approx. 0.5 ms; using 100 % wet on the output).
Finally, use a chorus at the end of the chain for some frequency modulation. This prevents the signal from phasing when it returns to the original audio track. I recommend that you check out some different settings to address your specific needs. Nevertheless, there are no rules.

Please listen to Sample1. It automatically starts with the original drum track. Then, after a few bars add the brightness send bus and switch it on and off every few seconds. While listening to the sample, focus on the hihat and the snare drum (also the overheads) since this is the frequency area where the effect works.

Punch: This is a very simple and commonly used trick in rock/pop audio productions. But it is as simple as effective. Therefore, make sure you have it at your disposal. Simply create a new send bus and use a compressor (try a short attack time and a very high ratio) on this track as an insert effect. I prefer to use the 1176SE for that purpose. The trick is to really compress almost the whole dynamic range out of the drums! As you can see I’m using the ratio in “all mode” and at a very high input level.

Listen to Sample2. Again, I am switching between the original track and another version with the punch send bus mixed in every few seconds. Obviously, the second version is a bit louder because of the added signal from the send bus. You might tend to like this version better because of the loudness, but the loudness is not the point in this case. I really want you to focus on the punch and aggressiveness.

Summary

Finding the right balance between the original audio signal and the returned signal from the send busses is your discretion. There are no specific guidelines except: What sounds good is good!

Let us see how our drum set sounds like with and without both send busses. Again, I’m switching the sends on and off every few seconds. You can listen to that comparison in Sample3.

Last but not least, I would like to show you the drum track in context. I did two mixdowns of the whole session to show you an A/B comparison of what it sounds like with and without the support of the send busses on the drum track (listen to sample4). See how the send busses help the drum track come out on top?

Note: As you might have guessed I am not using the send busses on the drum track exclusively. Like what I have mentioned before, it is also a stylistic device to put different instruments all together (similar to reverb). That means, I am using them on almost every audio track in my session (with different amounts). This naturally results in a smoother overall mix because of a similar sonic behavior in various instruments.

Guitars in Rock and Metal Soundtracks

Mon, 16 May 2011 18:44:40 +0000

Requirements

The most important thing is to have a guitar player who can deliver a solid performance. A poorly played performance can be corrected in the audio editing process but this is going to take a huge amount of precious time so if you have to meet a certain deadline you cannot afford to spend a lot of your time correcting wrong notes, timing problems, and so on. If you cannot afford to pay for a professional session player, and you are a moderate guitar player yourself, then try this:

If you are using guitar modeling software, such as Guitar Rig or Amplitube, you will want to record the signal dry and clean as a start. Remember, since the final guitar sound is created after using a plug-in you don’t have to record your guitar part in real-time. In fact, you can even use half-time or any slower tempo that you feel comfortable with. Then use the time-stretch function of your DAW (digital audio workstation) to make the tempo of your recording match the tempo of your project. Although this procedure does create solid results, there are some limitations. The time-stretch function either adds samples to your recording (according to a specific algorithm you usually need to select first), or it removes samples from your recording so artifacts will usually be audible if you use this function too much. Therefore, using less time-stretch leads to better results. The best quality you can get is a performance that is played in real-time with no time-stretching at all.

The Recording Process

Nine times out of ten, one single guitar track doesn’t lead to good sounding results, except maybe for a lead guitar solo. So it will be necessary to record each guitar part at least twice in order to double all the parts in the mix. Use different amp simulations for every guitar track and make sure that the sounds of the various amps complement each other. This means that you don’t want the doubled track to sound too much like the first one.

Note: If you just duplicate a recording the result is not doubling because this only adds volume to the original track (when using no frequency modulation effects).

And remember: You may use a slower tempo and/or you may record every part at a different time. If you do so, always start recording some bars before the actual part begins because you want to create a fluent transition from one recording to the other.

How To Place Guitars In A Mix

Most rock songs contain at least of two different guitars playing at the same time. The first thing we need to deal with is panning. Since we want to create a wide spread and powerful guitar wall, here is one way to go:

Guitar 1: approximately halfway to the right, doubled with another part playing exactly the same pattern panned completely to the left.
Guitar 2: approximately halfway to the left, doubled with another part playing exactly the same pattern panned completely to the right.
Guitar 3: in the middle playing higher notes which complement the chord structure.

Of course, all guitars may play the same part in unison or in octaves. If your song is complete with two guitars, then don’t use the third one. Whatever arrangement fits your needs best, go with it. Listen to example1 in order to get an idea of what the above described panning sounds like.

The next part we are going to focus on is equalization (EQ). Here are some basic guidelines for equalizing (EQing) guitars:

Do a low-cut at around 150Hz, because all frequencies below 150Hz belong to either the electric bass or the kick-drum but not to the guitar.
There are often some resonance frequencies between 180Hz and 500Hz, so find these by sweeping through these frequencies while reducing them between 5dB to 8dB. Always use the highest Q-factor possible during this entire process.
Take a look at the frequency spectrum between 2.7kHz and 3.6kHz, you will usually find one or more frequencies that are quite disturbing. If so, then sweep through the frequency spectrum again and reduce this frequency generously.
Often a smooth boost with a high-shelf at around 4kHz will brighten up the sound.

Finding the right frequencies requires experience or the trial and error method (which leads to experience anyway). Just don’t get frustrated too soon, because like everything else it’s a learning process. Listen to example2: the first version is without any equalization, the second version has every track equalized.

Reverb is very important for creating depth in the mix. For music genres like rock or metal the function of reverb is to emulate the environment of the recording studio as well as the instruments being played in that studio. Although there are no rules on how to use reverb, here are some points to think about:

Reverb normally is used as a send effect, not as an insert effect.
There are three things that define a space or room: the direct signal, early reflections (ER) and the tail, and all of them are equally important.
Reverb glues together different guitar tracks or sounds in general.
To find the appropriate length for the tail, just follow this guideline: the faster the song the shorter the tail.
By using reverb you can create a sonic foreground, middle, and background; rock guitars normally play in the middle of a mix, behind the vocals or the solo guitar but in front of the drums; I suggest playing with the arrival time and the amount of early reflections (ER) in order to position the guitars properly.

Here is an illustration of my reverb settings, which are quite usable for rock. The dry/wet-ratio should be at 100% because we are using the plug-in on a send bus. Additionally, you want the early reflections to arrive at the listener first; the tail comes later. In order to control the arrival time of the tail, we set the knob called “delay” to 21msec., and a little boost at around 4kHz will brighten up the overall sound.

In order to highlight a guitar solo use less reverb because this is equal to a higher amount of direct signal. This is going to push the lead guitar in front of the riff guitars that are playing in the middle of the mix. Now listen to example3: the first version is without additional reverb, the second version has Altiverb turned on. What reverb to use always depends upon the musical genre and upon your personal taste such as: plate, room, ambiance, spring, hall, etc. If you don’t have any specific ideas in mind, listen to bands in the desired musical genre and try to get as close to that as possible.

Additional Effects

You can add almost every effect to guitars like flanger, chorus, delay, wha-wha, octave-creators, etc.; however, at this point I would like to focus on how you can get a more aggressive sound.

If your guitars sound like the player was a bit lazy during the recording session, here is a little trick that you can use to sharpen the attacks later: Use a limiting amplifier (for example, the 1176SE from Universal Audio) as an insert effect on a separate send bus. Mix this bus slightly underneath your guitar tracks. This is going to add a bit more punch to the guitars. By the way, this trick is normally used to make drums sound more aggressive, but it also works for guitars.

Example4 shows the effect that the 1176SE adds to the guitars in an A to B comparison. It shouldn’t be very hard to tell which version is using the effect. Furthermore, listen to example5 since it shows the final mix of drums, bass and guitars (including a solo passage).

Note: The best effects are barely audible when active; however, when they are on bypass you can tell the difference in a second. So, because you want to retain a natural sound, don’t overdo it.

Shaping A Note

Mon, 09 May 2011 15:27:50 +0000

It is very important to keep in mind that we are trying to simulate humans, nothing else. Before we get down to business it might be helpful if you listen to some live performances of real instruments (e.g. on Youtube) since live performances are the source that we want to emulate.

The second part of this tutorial will show you how to apply this device in order to create real sounding instrument sections. For example, if you are going to build a 3-part trumpet section, there are some nuances that have the ability to reveal the part as being fake within just seconds. Our ears, and also your clients’ ears, are very experienced in recognizing how real instruments sound, so unfortunately only a few tiny changes or mistakes could be enough of a reason to burst the bubble and reveal the truth.

The 3 Parts Of A Note

Independent from its context a single note normally follows a particular curve. All notes produced on natural instruments follow this curve in some way.

To be sure, the three elements of the curve may vary from context to context in length and intensity, but do you realize that there are no parts of the note that are totally flat?

(1) The first segment obviously is a rise in intensity produced by a bow stroke on stringed instruments or the vibration of the air column in brass instruments. This segment may be very short, for example, when you listen to staccato phrases.

(2) The second segment is important because it enables the ability to distinguish between different dynamic layers – from pp to ff. The higher the peak the louder the note. Of course, you can hold a peak for a specific period of time but be aware of the fact that it does require some effort to allow a note to sound very loud for only a few seconds. This effort means that the bow of a string player needs more pressure and speed, and a wind player needs a lot more air.

(3) The last segment is the decay. When you play a single note it will usually become softer and quieter as it ends. So the decay is dynamic. Don’t forget that it can also be a crescendo into a stab – like the brass section in a big band. The best way to decide whether or not a line that consists of many notes can be played by a wind player is to whistle it yourself – but make sure that you pay attention to the dynamic markings. If you need to breathe before the line is over, you better include a break. This is one of the most important points you need to keep in mind in order to make your samples sound real. Additionally, when you write staccato or spiccato notes you don’t need a decay for your notes. Most of the samples come with release recordings so in most cases you don’t need to do anything manually.

Important: You must find a comfortable way to increase or decrease the intensity of your samples smoothly. In most cases the mod-wheel (CC#1) is the trigger to cross-fade between different dynamic layers by default, but it really depends on how the virtual instrument is set up and programmed. Don’t try to affect the intensity with volume because when you play an instrument from pp to ff its frequency spectrum changes. If you want an instrument to play very quietly, don’t use fortissimo samples and simply turn down the volume. That is not going to work! Therefore, the only thing that you affect by changing the volume is the balance and the relation to other instruments in your arrangement.

Note: In order to make a single note more interesting don’t forget to use the right amount of vibrato. In many cases a long decay is the perfect area for that modulation. Again, listen to some real players and understand when and how they use vibrato.

Examples Of Note Shaping On A Single Line

To begin, let’s listen to some examples in order to make clear what note shaping really is all about. The first part of Example1 is a short line taken from the German Christmas carol “Snow falls softly at Night” (original title: “Leise rieselt der Schnee”) played by a virtual alto saxophone without any note shaping. You will realize within seconds that this line is not played by a real alto sax.

Now, let’s add some CCs in order to make those notes sound better. In fact, there are several parameters that are used in order to produce a realistic performance. Here is a picture of the key editor. The line above is the melody line, the curves underneath depict the motions of the different CCs.

First, did you notice that the notes are not quantized to the grid. They vary slightly and therefore, mirror the behavior of a human player. The first line underneath the notes depicts the shape and motion within the notes. This is the subject of the preceding chapter. The blue curve underneath is the amount of vibrato. The peaks of the vibrato always fall into the decay of a note – at least in this case. Right at the bottom you will find the pitch bend, which plays around a perfect pitch. This is a real nuance, but it is natural and therefore, it adds another little bit of realism to the phrase. Briefly, here are some suggestions to keep in mind in order to make a midi performance sound more natural:

Do not quantize your notes to a grid.
Let every single note breathe; this is called tone shaping.
Do not forget about vibrato.
Do not use perfect pitches, use the pitch bend.
Do not copy and paste CC data.
Do not forget that wind players need to breathe.
Listen to live performances for references.

Note: In order to make a midi performance sound really good and very realistic you will need to edit many different parameters. However, on the other hand, if only one parameter is set a bit untypical, your performance will sound fake. So in this case, it is totally true: The devil is in the details!

Building Up A Section

A section consists of single lines that are played simultaneously. So the first step is to check whether or not the performance of every line is OK by itself. Next, there are some very important points you need to keep in mind.

Do not let all the instruments start at the same time. Instead, try to “roll” into the chord or notes.
Do not change the number of players during a song.
Use some small variations at the peak of every instrument.
Use different lengths for the decay.
Give every instrument a unique shape that refers to the decay.
Use different vibrato settings for every instrument.
Keep the section dynamic and moving along with regard to expression.

Please listen to Example2. The setting consists of two alto saxophones, one tenor saxophone and one baritone saxophone. The first alto takes the melody, and the other instruments support the main line by harmonization. The result is a four part section which moves block-wise. Again, the first snippet is a record of the lines without any note shaping; the second one uses expression, vibrato and pitch bend.

Summary

Your whole phrase is only as good as the quality of your single notes, so build up your phrases by shaping every note. This will bring dynamic, character realism and life into your performance. Don’t forget that a section is played by many different people, it is not the same person on all the instruments. So it is very important to vary the articulations, the decay and some other nuances (midi data in general) in order to retain the illusion that it is a live performance played by many musicians.

Note: Some people out there may consider doing all this as far too much effort. The truth is that real musicians spend many years sometimes decades, in fact, practicing to master a specific instrument. Since this also is a matter of training and repetition you will develop solid skills in midi editing in just a short period of time. Take your time to figure out what midi curves fit best into your arrangement and most importantly, have fun!

For those of you who are interested in listening to the finished song:

Compression Masterclass (Part 1)

Sat, 30 Apr 2011 14:15:00 +0000

In many of the articles I’ve read and wrote myself, the settings and the general theory of compression are prioritized rather than focusing on the best way to use compression/limiting in a certain situation, or even if you should be using it at all. As a result producers understand why they should be using compression but don't actually know when and how they should be applying it.

To help clear this up, in this tutorial I will identify three possible situations you may face, explain how or if you would apply compression/limiting and the affect the process will have on your sound.

Practical Compression: altering the sound of your drums

Today, compression is not only used for narrowing the difference between high and low audio levels, but is used as an effect to accentuate different aspects of a waveform. During the record stage, you may have inserted a compressor to aid in controlling your drum kit but to enhance the tones of your drums it is worthwhile using practical compression on individual elements during the mixing stage. Okay let’s focus on the main element of your drum kit, the kick drum. Initially you want to be asking yourself what elements of the kick drum you want to emphasize and then go through each parameter of the compressor, keeping in mind what effect you want.

Sustaining the Kick Drum

The most common way of using practical compression on a kick drum is to alter how the sound decays. A kick drum that remains at full volume throughout the note is going to be considerably louder than one where the transients hits and then fades out. So what we need a compressor to do is clamp on the initial transients so they dip in relation to the rest of the kick drum, therefore increasing the perceived loudness of the decay.

To do this we will need to bring the threshold down to a level where only the initial transients are being affected by the compressor and set an average ratio of 4:1, however adjust this setting if you feel the transients are being squashed or you feel there needs to be added compression. Subsequent to this we need to use the attack and release settings, which I believe are the compressors most influential parameter, to control how quickly the transients are being compressed and how quickly the compressor returns to unity gain. We need to use a fairly quick attack time, so to clamp on the transients immediately however too fast of an attack time will reduce the gain of the transients too quickly. We also need to use a quick release time to allow the compressor to reset after the initial hit and although ‘gain pumping’ like this is often frowned upon it can be a good technique to increase the loudness of your kick drum.

Be cautious with fast release times though as you don’t want the ‘gain pumping’ to be too evident. Finally use the make-up gain to compensate for the gain reduction.

Focusing on the transients

Practical compression can also be used to highlight the transients of a kick drum. This could be down to personal preference, genre, or just wanting the high end of the kick drum to be more apparent in the mix. Similar to the previous compression settings the main parameters that significantly alter the characteristics of the kick drum are the attack and the release settings. If we keep a similar release time but slower the attack time slightly, more of the transients will be allowed through therefore they are emphasized in relation to the rest of the waveform.

Controlling Pianos

How you apply compression on a piano is wholly dependent on the unevenness of volume, the tempo and how busy the track is, so before inserting a compressor you need to categorize the piece so you can highlight the problems you need to put right with compression.

So you’ve now categorized your track; Is it slow? Is it fast? Is it percussive? Is the pianist comping? We now need to strap a compressor on the piano bus and control it depending on the category you’ve given it.

Okay, now that you have inserted a compressor onto the piano bus I will run through several types of piano pieces and give general guidelines on how you should be applying compression.

A fast track played with a percussive touch

Due to the quick, percussive nature of this type of track we need to apply a compressor to clamp down hard on the transients to gain a bit more control, so the piece fits better with the rest of the mix. To ensure that the compressor is only affecting the peaks, set the threshold so that is just above the average signal level or if you only want to catch the largest peaks, set the threshold so that only those peaks alone affect the gain reduction meter. With your threshold set, you now need to decide how firmly you want your compressor to respond to transients passing over the threshold.

If there are only a few real problematic spikes you can use a hard ratio setting or even a limiter to help considerably reduce their gain however if there a lot of small spikes you are far better off setting a ratio of 3-4:1 to control them in a more naturally sounding way. As far as attack and release settings go, again you need to set them depending on the nature of the track. If there only a few large peaks set a very fast attack and release time so that you are only affecting those peaks and restraining them from overshooting the threshold.

Part 2 coming soon...

Andrew Charlton owns Gold Shore Records an independent record label and FL Studio Tutorial Resource

Pimp My Orchestra - High Strings

Wed, 27 Apr 2011 16:02:10 +0000

Introduction

Keep in mind that there no rules and no limitations, just one doctrine: If it sounds great, it is great!

Note: It’s important to know how to write lines for strings in order to make them sound natural and sophisticated in the first place. No audio editing trick will turn a poorly written passage into something beautiful, so it may be a good idea to develop solid skills in both these areas – writing lines and audio editing.

Now here is a good and easy way to set up a spot on bow strokes, bow movements and nuances in order to freshen up the overall sound. The method described was originally used to brighten up vocal recordings, but it does a great job on strings as well.

Adding “air” and sizzle to high strings

Instead of editing the high strings audio track itself using insert effects, I am going to set up a separate send track or send bus. This way I am creating an option to edit the sound by adding new information to the original track without touching the original audio data at all. You will need to load three insert effects on the send bus in this particular order:

Equalizer (EQ)
Short ambience (Reverb)
Chorus

The following illustration shows the flow of the audio signal in your sequencer so that you will have an idea of what is going on. The blue line represents the original audio signal, and the red line represents the added send-return signal. As you can see, the output of the send bus returns and adds to the output of the original audio track.

Now let’s have a closer look at the settings of each of the effects. Obviously, every effect has its own relevance in order to create the sizzle that we are looking for. The first effect in the chain is the EQ.

By using an EQ, we can determine the area of frequencies that we want to manipulate. As you can see, there is a high pass at around 4kHz. It is obvious that we want to get rid of all frequencies below 4kHz, so in the next steps only the high frequencies will be edited.

It’s important to use a short period of time for the tail because otherwise your overall sound will become very nontransparent later. In our case, the main purpose of the reverb is to create a slightly longer signal so the chorus gets “something” to work on. It is also possible to experiment with the settings for early reflections (ER) in order to push the sound a little more backwards, just be sure that your reverb output is set to 100% wet.

A chorus does permanent frequency modulation. So because we are going to return this signal back to our original audio track, the chorus prevents us from phasing. If you added up the same signal twice phasing would be the natural result because of the identical wave forms of the audio signal. Thus, by using a chorus, we generate a completely new wave form, and therefore, it is safe to add this bus signal to our original signal without causing any phasing problems.

An example for reference

After all three effects are set up on the bus, we are ready to turn this process into action. Listen to example1, which shows how a high string passage with the send bus turned off will sound. This means that you are listening to the original audio signal only. Example2 is a rendering of both, the original audio signal as well as the send bus audio signal. It should be obvious to you that the sound is much brighter and more open. For demonstration purposes only, the effect is overdone, so make sure that you use it more subtly on your own productions. Example3 is a little snippet of the final mix that is supported by drums and an upright bass.

Note: In order to find the appropriate level of volume for the send bus, here is some (hopefully) helpful advice on that. Like it is so often mentioned, the most important thing is that you don’t overuse the effect. It should be barely audible in the mix, but it should be very clear when it’s switched off in an A to B comparison.

Of course, it’s a very important creative decision, whether or not this effect should be used at all in a particular project. However, it can certainly be very helpful to know about this method when someone asks you to give your strings more air and brightness.

Orchestral Panning for Realism

Tue, 05 Oct 2010 17:01:52 +0000

Even if you have the best samples and have set up your reverb well, you still have some work to do. Luckily, it’s quite simple and can make a significant difference in how real your mix sounds. The missing link is panning, and this tutorial will teach you to be aware of how to use it to your advantage.

What Is Panning?

If we consider reverb to be a means of changing the depth of a mix by simulating distance between the listener and the instruments, then we can think of panning as the way we change the horizontal positioning of the instruments in the mix.

You likely recognize it from mixes where it was used as a special effect, where an element pans from left to right and back again, giving a sense of motion:

But there’s more to it than just flashy special effects.

Avoiding the Organ

Panning is a deceptively simple technique that tends to be overlooked by many media composers. Despite the advancement of sampling technology, trying to render an orchestral tutti with everything centered will produce the dreaded “organ effect”; a complex interference artifact that results in several frequency ranges being strangely boosted and distorted:

The effect is most obvious when contrasted against a properly panned alternative:

Don’t worry if you didn’t hear it at first: panning is one of the more subtle influences on our perception of realism and it mostly acts subliminally unless we’re paying attention or it’s used in an extreme manner. Here we can notice the most significant difference between a properly panned mix and a poorly panned one: clarity.

The initial version has all the instruments crunched in the center, so even though they sound like they’re playing in a hall, it’s as if they’re nothing but a giant totem pole of brass instruments. In other words, if you close your eyes, you’re not hearing individual sections so much as one integrated wall of sound. The second version has the instruments panned appropriately so that even though it’s a tutti, you can still clearly distinguish between the trumpets, horns, trombones, and tuba.

To demonstrate the importance of this clarity, we need to look at a different type of example.

Polyphonic Clarity

It’s all well and good to be able to differentiate between sections in a tutti segment, but where panning can really help elevate the realism of a mix is in the more agitated sections of a composition.

Let’s take a short passage for strings only. Here is the initial version first:

When the sections are playing independently, it sounds pretty good, but you’ll notice in the final section where they’re all playing together that the frequencies start to stack a bit strangely and you get an unfocused sound. Let’s hear it with the strings all panned according to usual orchestral seating:

This time, we can clearly distinguish between the violins, violas, celli, and basses and when they’re all playing together the effect is a more defined sound.

What Goes Where?

You can, of course, pan your instruments wherever you want them in a mix; and indeed some genres of music require very specific panning to sound legitimate. The symphony orchestra tends to be among the trickiest in terms of instrument placement simply because there are so many instruments and the arrangement of an orchestra varies depending on several factors.

For instance, there is no real “standard” when it comes to the arrangement of a modern orchestra. Broadly, the strings go in front, followed by the woodwinds and brass, and finally percussion in the back, but where each individual section is placed horizontally depends on whether it’s a large symphony orchestra or a small film score orchestra.

For reference though, let’s take a look at a basic seating chart:

This chart is a useful reference, but it shouldn’t be taken as a one-stop solution. For instance, if you’re sequencing a piece with a prominent solo part, you may want to have the soloist panned closer to the center to be clearer and louder. Likewise, you may choose to deviate from typical seating arrangements if you’re looking for a unique or unfamiliar texture.

How Far?

Panning is as much an art form as the rest of mixing, and deciding how far to pan your instruments is not just a matter of technicality but also of taste.

When crafting the illusion of realism, you can also use panning to augment your reverb setup for creating a sense of depth. If you think about it, assuming we’re imagining an orchestra playing in a hall, the further back you are from the stage, the less defined the sound is and the closer everything sounds to the center. If you’re looking for a very clear sound then you’d probably sit closer, in which case the horizontal positioning of the instruments would be much more obvious.

In this way, you can use panning to help simulate sitting closer or further from the ensemble while maintaining the sense of hall size with your reverb.

Parting Thoughts

While panning is a necessary and subtle tool for polishing up the realism of a track, you have to be careful with it because over-doing the panning can result in strange things in certain listening scenarios. For example, while your track may sound great with a certain wide panning in stereo, it may well be that listening to it on a mono system — a phone, cheap computer/laptop speakers, etc. — will cause the balance of levels to be way off and distort the intended sound.

To ward off such problems, always check your mixes on a few different systems of different quality — whether or not you’ve done anything special with panning. Remember, the easy part of mixing is getting it to sound good on your system; the difficult part is getting it to sound good on almost any system.

That being said, try experimenting with how you pan elements of your track and see what results you get. It can be a subtle effect, but the results are definitely noticeable if well implemented!

Creating Cinematic Soundscapes With Convolution Reverb

Fri, 17 Sep 2010 12:56:14 +0000

In this tutorial, we’ll examine one of the most flexible and accessible methods for producing compelling ambient sounds to use in your compositions: by using convolution reverb as a sound design tool. But first…

What Is Convolution Reverb?

It seems safe to assume that most folks with an interest in digital music production will have at least a passing familiarity with the concept of convolution reverb. For everyone else, the term refers to a specific type of reverb plugin that makes use of so-called impulse responses to precisely replicate the sonic behavior of echoes in a particular space.

An impulse response is typically a recording of a gunshot, a sine sweep, or a similar basic sound from which the convolution plugin can then extrapolate all the information it needs. The waveforms typically look like the one below, with a strong attack at beginning and then a sloping decay as the echoes die out. Once you pass a signal through the plugin, it uses the impulse response to attempt a convincing recreation of the way the sound would behave in the given chamber.

Most readers will be familiar with this basic usage of convolution reverb and will use several standard impulse responses to replicate the acoustics of a concert hall (for instance). This would allow you to go from an unprocessed, or dry, French horn sound:

To this much more realistic version:

And as the images indicate (this is simply Kontakt’s built-in convolution effect, nothing fancy), all we did was load up a basic concert hall impulse response! Perhaps the best part about this system is that most major DAWs and samplers come with their own built-in convolution reverb effects, so chances are you already have everything you need to make use of it. And there’s more to it than meets the eye…

Beyond Reverb

Despite its usefulness in creating a realistic mix, convolution reverb plugins are also extremely versatile tools for unconventional sound processing. Because you can capture the sound of virtually any physical space, convolution reverb allows you to digitally produce sonic environments that are simply impossible to create in the real world. How would a symphony sound playing inside your piano? How about a brass quintet sitting in your flower pot? Or underwater?

This often-overlooked fact allows for endless creative potential: you can record your own impulse responses and use them to give your music a distinctly personal sound. The possibilities are limited only by your imagination. But we want to take the process even further off the beaten path to explore its potential for designing ambient sounds, so without further ado:

Crafting a Convoluted Ambience

Organic Sources

Since there is no rule that says you must use a specifically engineered impulse response in the plugin, commercial vendors have recently been catching on to the fact that, for the most part, you can load any standard .WAV file into most convolution plugins. This means that instead of just the basic impulse response of a concert hall…

You can break out a bizarre sound as your impulse response and achieve surprising sounds from the same French horn patch by tweaking the signal mix so that there’s significantly more of the wet signal than the dry — that is, more of the “reverb” sound than the original — and seeing what happens:

Suddenly, you can hear the overtones shimmering and everything washes together like a synth pad.

Synthetic Sources

1. The Raw Sound

And speaking of synths, some of your best results are likely to come from starting with a synth as your sound source. Anything from a basic lead to a growling bass to a pad will lend itself spectacularly well to convolution designing. To illustrate the point, let’s take a basic FM synthesis percussion loop and listen to it raw for starters:

2. Baby Steps (Backwards)

Funky, but since we’re after ambient sounds for this tutorial, let’s see what we can do to turn it into a useful cinematic element. For starters, let’s take this opportunity to showcase another neat manipulation that can be performed using nothing but a standard hall impulse response. Just like there’s no rule saying you must use an typical impulse response, there’s also nothing stopping you from importing your basic impulse response in a strange way — like backwards!

Taking a generic impulse response and reversing it in your favourite audio editor will yield an interesting result. Notice how now the sharp attack of the gunshot is replaced by a soft build to an abrupt drop-off? Well it just so happens that our percussive loop is full of sharp attacks that we’d like to get rid of to try and smooth it out into an ambient sound, so let’s run it through this impulse response and see what we get:

3. Creating the Chain

The effect is not bad, and the swishing high frequencies are great, but it’s still a bit too rhythmic and defined overall, so we can’t stop yet.

Since we have a useful starting point, we need not go back to the drawing board. As with any other type of plugin, convolution plugin effects can be stacked so that one feeds into the next and you end up with a chain of processing. This allows for powerful and drastic changes to be wrought and will enable us to put the finishing touches on our sound.

Let’s continue by selecting a second impulse response. I’ve chosen a sound that has a fairly soft attack and a long, sustained decay so that we get a smooth, stretched-out sound. Passing the original loop through it yields the following:

4. Finishing Touches

This on its own is already a cool sound, but it feels a bit prickly and lacks the kind of sweeping modulation that makes a useful ambient effect. However, if we first run the loop through the reversed impulse response and then through this one, we finally hit the sweet spot:

Those sweeping high frequencies are now less punctuated, so we can easily achieve a patient and dramatic build with plenty of subtle sonic activity in the overtones.

Of course, at this point we could continue on and add various EQ and other effects to our processing chain to achieve an even more distinctive sound, but even without that we have used the power of convolution to design a couple of useful ambient sounds using nothing but a basic French horn patch and a synth percussion loop!

This, of course, is merely one way in which you can use the tools you have to generate fresh and compelling material to use in your next project. Never underestimate the power of your DAW’s built-in plugins and some ingenuity!