Samplers - part 3 (Dr. REX)

Aright. So we've gotten pretty well into using samplers in Reason, specifically the NN19 and NNXT. We've used them to:

• Play back multisample patches of instruments in Reason.
  
• Load our own samples and map them to their own keyzones.
• Loop a drum sample.
• Chop up a sample and play the individual drum hits with a keyboard.
  

So we've covered most of the common features that are used in creating sample-based music, except for one big one:

TIMESTRETCHING

What is "timestretching"?

Wikipedia:
"Timestretching is the process of changing the speed or duration of an audio signal without affecting its pitch."

This is most useful when you find the perfect sounding sample loop that isn't quite at the right tempo.

Here's an example:

Say you make a song that is at 89 BPMs. Perfect tempo, you can't stop nodding your head when you listen to the song. Then you find this great drum loop that you would be perfect for fitting into the song, but its tempo is 97 BPMs.

Now, you know that by decreasing the pitch, you can make the tempo slower, so you drop the pitch and the drums are now playing exactly in time with the song. BUT, when you listen to it, the drums are all kind of screwed up and psychedelic, and they don't sound anything like the clean drum sample you started with. The kick drum is not thumping, the snare isn't cracking, the hi hats aren't crisp, so there's really no point in using it anymore.

But what if you were able to make the loop fit the song, without changing the pitch? That would change everything, right? Welcome to the wonderful world of timestretching, my friends.

In terms of hip hop history, timestretching first started getting used a lot with the release of the Akai MPC2000XL. That was the first beat machine that incorporated the technology, and it became a pretty standard tool after that. Changed the production game.

Problem: Reason does not do timestretching!

So, why am I even bothering to tell you about it then?

For one thing, most DAWs (Pro Tools, Logic, Cubase, etc.) have a timestretching function, so you should be aware of it for when we get into using Pro Tools. But also, although it doesn't do timestretching per se, Reason does use a similar technology to accomplish a similar thing. In addition to the NN19 and NNXT, Reason has a sample-playback instrument called the Dr. REX.

Big difference between the Dr. REX and the other samplers is that while the NN19/NNXT can play back standard audio files (WAV, AIF), the Dr. REX can only playback special files created using another software program called Recycle.

Recycle (which is made by Propellerhead, the same company that makes Reason) basically allows you to chop an audio file up into slices. The program automatically creates slices at points called attack transients, which are basically the main points where drums or other sounds are hitting. Once you've got the file chopped up into slices, you can load it into the Dr. REX player and it will automatically adjust the sample to whatever tempo the song is set to.


One thing that you should be aware of is that neither timestretching, nor beat slicing is a perfect science. The farther you take a sample away from its original tempo, the less realistic it's going to sound.

That said, let's work check this out:

1. Go into Media Share>Class Materials>Samples and find the file called Funky Drummer_break.rx2. Copy it into your folder on your hard drive. Be sure that you are copying the REX 2 file, not the regular wav file!
2. Create a new song in Reason and create a Dr. REX sample player.
3. Load the Funky Drummer_break sample into the Dr. REX.
4. Play some keys on your keyboard. What do you hear? Does this remind you of anything?
5. Set the tempo to 106 BPMs.
6. Set your loop markers for four bars (L on Bar 1, R on Bar 5)
7. Click on the button that says "To Track". What happens? What does this remind you of?
8. Now drop the tempo down to about 90 BPMs. What's it sounding like now?
9. Find the Transpose knob on the front of the sampler and change the key up to F#.
10. Transpose it back down to C.
11. Set it to a tempo that you like and incorporate it into a song. 24 bars. Use the filters and/or LFO controls.

Samplers- part 2 (loops)

Today we're going to get into using samplers to work with loops. Specifically, we're going to start working with breakbeats.

As we learned earlier, breakbeats were the parts of old school funk and soul records where all the instruments would drop out except for the drums. This is the part of the song where people would really freak out and go off dancing.

Kool Herc figured out that by having two copies of a record, he could extend the breakbeat for as long as he wanted, playing one record, then the other, then rewinding the break on the first record, etc. Some people would say that this was the real start of hip hop as a new, distinct style of music, using the breakbeat as the backbone of the whole genre.

A few years later, when samplers became affordable and had long enough sampling times (see "Samplers - part 1"), people were able to just record the breaks into the samplers and have them loop automatically for however long they wanted.

Today we're going to get into working with a particularly famous break from a song called "The Funky Drummer", by James Brown.

Let's listen to the original for a second...

Funky Drummer

So now we're going to work on getting the break into our own songs so that we can use it.

But first...

Disclaimer: Sampling other peoples' work without their permission is a violation of copyright laws. If you are merely sampling for your own educational purposes (like we are doing in this class), then you should be fine. BUT if you intend to create and SELL music that incorporates samples of other artists' work, you are legally required to get all necessary permissions from the owners for the use of those samples. Think about it...how would you feel about someone taking your music without your permission and then making money off it?

That said, check it out...

1. First, let's grab the sample you're going to be working with and copy it to your hard drive. Go to Media Share>Class Materials>Samples and find the file called "Funky Drummer_break.wav". Copy that into your folder on your hard drive.
2. Now create a new Reason song called "Breakbeat lesson 1_your name".
3. Create an NN19 sampler in your rack and make sure it is connected to a Mixer 14:2. 4. Click on the Browse Sample button at the top of the NN19.
5. Load the Funky Drummer_break file into the sampler.
6. Play some notes on your Oxygen 8 keyboard and find the key that plays the loop in its original pitch.
7. So what we've got here is a 1-bar drum break. In order to make it line up with everything else in your song, we've got to make the tempo of the song fit with the tempo of the break, so...first, set your L and R markers so that they are marking off a single measure (put the L on bar 1 and R on bar 2).
8. Now hit Shift-Tab on your keyboard to go into the Edit View of your NN19 sequencer track (the part that shows the keyboard and notes).
9. Click on the Pencil Tool.
10. Find key C3 and draw a note that is exactly one bar long.
11. Make sure the Loop On/Off button is lit up, then hit play and listen to the loop. What is happening?
12. Now let's adjust the tempo so that the sample loops perfectly...Lower the tempo until you get down to 101 BPMs. Sounds good, right?
13. OK, so now we've got something that we can work with.
14. Switch over to the Arrange View (hit Shift-Tab).
15. Now we're going to mark this sample loop into a group. Using the Pencil Tool, mark off 1-bar. You should now have a region that you can copy and paste as many times as you want in the song.

From here, you can go ahead and create a Redrum drum machine and start making a beat that can either go over the loop, or switch off parts with it. You might also want to try playing with the Filters on the NN19 and seeing how you can switch up the sound of the loop.

Your assignment:

Create a 24-Bar song that incorporates both the drum break and the your own original Redrum drum pattern. Here's the catch: the loop has to be played back at a pitch other than its original pitch, meaning a key other than C3. So you will probably have to adjust the tempo to make this work.

Hint: When you're trying to match the tempo, if you hear the loop getting cut off too soon, then the tempo is too fast. If you hear a gap at the end of the loop before it starts playing again, then the tempo is too slow.

{Multisample Lesson}

Now we are going to do a lesson where we load samples of our own voices into the NN19.

1. Record two vocal samples into Pro Tools (the instructor will help you with this).
2. Load your samples into the Media Share folder and copy files into your folder on your computer
3. Open Reason and create a new song.
4. Create a NN19 Sampler
5. In the screen of the NN19 right click and select 'Split Key Zone'.
6. Click on the Key Zone on the left side and click the browse sample button
7. Load one of you two samples
8. Now click on the Key Zone on the right side of your NN19 screen and click the Browse Sample button.
9. Play some of the keys on your keyboard and notice where the pitch changes are happening.
10. Now lets change the key where the sample on the right plays its original pitch.
11. Find the knob called Root Key (under the NN19 screen).
12. Turn the knob until the root key is C4.
13. Your two samples should now play in their original pitches on Keys C3 and C4.
14. Make a beat that incorporates your samples.

Samplers - part 1

So, we've covered synthesizers and worked with each of the different synths in Reason. Now besides synths, there is a second category of instrument that you have probably already experimented with: the sampler.

"What's a sampler," you ask?

Better question: what's a sample?

There are actually two different definitions, depending on who you're talking to. You need to know both:

1. (common definition) A piece of a sound recording that is taken and incorporated into a new musical composition.

2. (technical definition) The smallest unit of a digital audio recording.

Let me just deal with the second one, briefly, because this is at the heart of understanding digital audio. We'll get deeper into this when we start working with Pro Tools and recording stuff.

OK, so we know that sound is really the vibration of air molecules, and that it moves in waves, right? So, when you've got sound going into a device like a microphone (aka a transducer), what's happening is that that sound wave is being converted into an electrical signal that has all the same properties (frequency and amplitude!) as the wave that was traveling in the air. This is called analog audio, because the electric wave is analogous to the sound wave in the air. Stay with me for a minute...

So, when you're talking about an analog audio recording, you're really talking about an actual physical representation of a sound wave that exists in one form or another. With digital audio on the other hand, what's happening is a computer is taking tons of tiny snapshots of the sound wave, and storing that information as a zeros and ones. The waveform doesn't physically exist anywhere, there's just a bunch of computer datat that describes what the wave is supposed to look and sound like. Don't stress about this, we'll get into this later, right now, just understand this:

The smallest snapshot that a computer takes of a sound wave is called a sample.

So, when you're listening to any kind of digital audio (MP3, CD, etc.), what you're actually listening to is one big, long series of these samples. "How many samples, you ask?" Well, I'm glad you asked! It depends on what format you're talking about, but...

CD quality audio samples at a rate of 44.1 kHz

Or,

44,100 samples per second.

Basically, the higher the sample rate, the better the quality of your audio. Most professional recording is done at 48 kHz or 96 kHz.

So we know what a sample is. Getting back to the original question then, what is a sampler? Wikipedia says:

A sampler is an electronic music instrument closely related to a synthesizer. Instead of generating sounds from scratch, however, a sampler starts with multiple recordings (or “samples”) of different sounds, and then plays each back based on how the instrument is configured.

So, a sampler is basically an instrument that can play back digital audio recordings (aka "samples").

Here's a quick and dirty history of hardware samplers:

Synclavier

• Came out in 1975
• Cost $200-300k

Fairlight CMI

• Came out in 1983
• Cost about $25,000
• Supposedly Took 2 hours to boot up
• Featured basic MIDI capability
• Used by Herbie Hancock ("Rockit") and Kurtis Blow ("If I Ruled the World")

E-mu Emulator II

• Came out in 1984
• Cost $8,000 - 10,000
• Had an option to add a hard drive

E-mu SP12

• Came out in 1985
• Had 24 onboard drum sounds
• Featured 1.2 seconds of sample time
• Had MIDI sequencing, which allowed you to create whole songs
  
• One of the original hip hop production tools - classic gritty sound
  
• Replaced by the SP-1200 (10 sec. sample time)
  

Akai MPC60

• Came out in 1988
• 13.1 sec of sample time (upgradable to 26.2)
  
• 16 velocity sensitive pads
• 40 kHz sample rate
• Gave birth to the whole MPC family (2000/3000/2000XL), which became the standard hip hop production tools throughout the 90s.
• Used by DJ Shadow on Endtroducing - the first album created only with samples.
  

Ensoniq ASR-10

• Came out in 1992
• Up to 44.1 kHz sample rate (CD quality!)
• featured lots of onboard FX and sound processing
• Not as popular as Akai MPCs, but it's still widely used by producers for its unique sound (Timbaland, Neptunes, Kanye).

So, from a hip hop/electronic music perspective, the sampler was a huge breakthrough because it allowed you to capture a drum break (or other sound) and then loop it, creating an infinite beat. (We'll get into this more in the next lesson).

But loops weren't the only thing samplers were good for.

Think about it, you have a machine that can take a recording of ANY sound and then play it back. That's cool, but what's really amazing is that you could actually hook up a tool like a keyboard to a sampler and then play the sample back, at all the different pitches of the keyboard. So, you could take a sample of a real instrument (say, a saxophone), or even a completely random sound, and then play it like a piano.

Only, it's never quite that simple...

See, the problem is, most samplers do know how to adjust the pitch of a sample so that it plays the correct pitches on all the keys of the keyboard, BUT the farther a sample gets away from its original pitch, the less it sounds like a real instrument.

Let's see for ourselves, using Reason's NN19 sampler:

• Open Reason, create a new song, save it as "Multisample demo"
• Make sure there is a Mixer 14:2 in your rack
• Create an NN19 sampler
• Right click on the Browse Patch button and click on Initialize Patch.
  
• Now click on the Browse Sample button
• Go to Reason Factory Sound Bank>NN19 Patches>Piano>Bright Piano>PianoBC3.wav and click OK to load that audio file.
• Play C3 (the first white key on the left side of the keyboard) and listen to how it sounds.
• Now play C4 (one octave up from C3). How does that sound?
• Now hit the Octave Down button on your keyboard and play the low C key again. How does it sound?

Now create a second NN19 in your rack.

• For this one, click on the Browse Patch button.
• Go to Reason Factory Sound Bank>NN19 Patches>Piano>BRIGHTPIANO.smp and click OK.
• Now play the same notes that you did a minute ago. How do they sound different?
  

So, most modern samplers use what are called multisamples for recreating real instruments. Multisamples are just a bunch of separate, individual audio files of an instrument being played at different pitches. These audio files are then mapped to certain sections of the keyboard that are close to the original pitch of the sample. The collections of these audio files and the information about where to map them on the keyboard are conveniently put into what are called patches. Patches make it easy for you to quickly load up a good sounding instrument into your sampler and start playing, without having to deal with mapping all the sounds yourself.

Really good quality multisamples are especially important when you're trying to realistically recreate things like orchestral instruments, which have highly detailed sounds and are easy to tell when they are off.

So...your assignment right now is to find some good orchestral patches and use them in a song.

Please make me a song that meets ALL of the following requirements:

• The song must be at least 24 bars long.
  
• You may only use 1 Redrum and as many NN19s as you want. NO OTHER INSTRUMENTS!
• 1 of the NN19s must be some kind of Strings
• 1 of the NN19s must be some kind of Brass
  
• 1 of the NN19s must be some kind of Woodwinds

Other than the above requirements, it's up to you. Please put your beat in today's folder in the Media Share folder.

Synthesizers-part 2 (Malstrom and Thor)

Alright, so last time we got into the knob-tastic world of synthesizers and did a little bit of programming our own patches using the Subtractor.

Today, we're just going to take a little bit more time to talk about synths and take a look at the other synth instruments that are available in Reason.

So, we are pretty familiar with the Subtractor. Do you remember what type of synthesizer the Subtractor is?

Analog synthesis (aka "subtractive synthesis") was the classic, old school way that synths worked. This method of synthesis used actual electrical signals as the basis of the sounds it made. Some of the most famous makers of early analog synths included Moog, ARP and Roland. Here is a picture of the Roland Juno 106, one of the first "affordable" synths on the market:


This is one of the all-time most used synth. It is one of the classic sounds from the 80's. You've probably heard it on hundreds of songs.

As technology progressed, instrument makers started using different methods of synthesis that allowed for more complex sounds. The next big one was called FM Synthesis, or Frequency Modulation Synthesis. Using this method, you could basically take two signals and have one mathematically alter the other. Don't worry, we're not going to get too deep into this, but just know that the sounds were more advanced and allowed people to try to create synth patches that resembled the sounds of instruments. That's when you started to hear cheesy synth horns and strings on pop songs.

After that, things started to get Digital. With digital synthesis, you had a lot more options and a number of synthesis methods came out. One of these was Granular Synthesis. With Granular Synthesis, you are basically taking a sample of an actual audio sample (say a flute sound), and then breaking it up into hundreds of little tiny snippets, or grains. Then, using the different controls, you can reconfigure and mess with all these grains and create completely orignal, crazy new sounds.

This brings us back to Reason:


The Malstrom pretty much uses Granular Synthesis as the way it creates its sounds, though it is laid out very similarly to the Subtractor. Like the Subtractor, it uses two oscillators at the starting point of its sounds. However, if you go through the different patterns, you see that it's not using simple waveforms, but the names of actual instruments as the basis of its sounds. Plus, using the Index sliders you can determine what part of the sample you want to work with.

The third and newest Reason synth is the Thor.

This guy is pretty intense and sounds AWESOME. Basically, instead of using two Oscillators to create its sounds, it uses 6. It also has 6 filters. This guy is incorporating several styles of synthesis at once and allowing for highly detailed and complex sounds.

So, start comparing these different synths and seeing the kinds of sounds they all make. Some are definitely better than others for different styles of music. But they're all tools available for you to use and combine to create your own original sounds.

Synthesizers - part 1 (Subtractor)

There are basically 2 categories of instrument in Reason. For the next few classes, we're going to take a look at the first one: synthesizers.


What is a synthesizer?

Wikipedia says:

A synthesizer is an electronic instrument capable of producing a wide variety of sounds by generating and combining signals of different frequencies. There are three main types of synthesizers, which differ in operation; analog, digital and software-based. Synthesizers create electrical signals, rather than direct sounds, which are then processed through a loudspeaker or set of headphones.

So, basically, you've got a device that electronically generates one or more audio signals. And then by combining and processing those signals, you can create completely orignal sounds. There are several ways that different types of synthesizers operate. The simplest one is called analog, or "subtractive synthesis".

We are actually already familiar with the sounds of subtractive synthesis though working with our friend...

...the Subtractor!

So, obviously there is a lot going on with all these knobs and sliders and stuff. But once you know how to look at it, it really isn't so overwhelming. Today we're just going to focus on three sections, and from there you should know a lot about almost every instrument in Reason. The three sections are:

1. Oscillators
2. Filters
3. ADSR Envelopes.

We'll start with the first: Oscillators.

Oscillators are basically the heart of the instrument. This is where the sounds originate from. The Subtractor has a handful of very basic types of sound waves that it uses as the raw material for creating instrument sounds. Think of these as the block of stone that a sculptor starts with before he/she starts chiseling it into a specific shape. Any guess what those sound waves might be?

That's right: Sine waves, Square waves, Triangle waves, Sawtooth waves, and some others that are variations of these guys.

Subtractor has two different oscillators that it can combine to create more complex sounds. Let's check this out for a second...(demonstration)

OK, so besides being able to play back two sound waves at the same time, Subtractor lets you adjust the pitch of each each one. What do you think that does to the sound? It also lets you add noise, if you're into that.

Next, we have the Filters.


So, if the sound waves are the block of stone that a sculptor starts with, then the filters are like the chisels and other tools that he/she uses to shape it into what he/she wants. Now filters are what put the "subtractive" into subtractive synthesis, and they are a really common audio production tool in general. Think about it, what does a filter do?

We've got four basic types of filters to choose from, and they all reject different parts of the frequency spectrum. They are:

• Low Pass (LP)
• High Pass (HP)
• Band Pass (BP)
• Band Stop (aka "Notch")

Remember, the key word here is "pass"; what frequencies are being allowed to pass through the filter? In a Low Pass filter, the "lows" are being allowed to "pass". In a High Pass filter, the "highs" are being allowed to "pass". Here, let me just show you... (demonstration).

So, in Subtractor, you can select whichever filter you want to work with by clicking on the red dot. Then you can adjust the Cutoff frequency (the point where the filter starts working) by dragging the slider called Freq.

OK, so the last section of the synth we're going to cover today is the ADSR Envelope.

ADSR just stands for: Attack, Decay, Sustain, Release, and it's referring to the way the volume of a sound evolves over time. Check out this picture of a waveform:

What it is showing is the different parts of the total sound. Briefly:

• Attack - the quick rise of the volume up to the highest level.
• Decay - the drop from the highest peak to the average level of the sound
• Sustain - the average level of the sound
• Release - the fade out

Here is a common diagram of an ADSR Envelope:

Now, every sound has these four basic qualities, but with synthesizers, you can actually control the points that these things are happening. You can do this in the section called Amp. What do you think "Amp" is short for?

OK, so if your mind isn't completely overloaded yet, know that you can actually use ADSR Envelopes for more than just the volume of the soundwave itself. One of the most common things is to connect it to the Filter and have the filter moving in a completely different way than the Amplitude envelope. Bottom line: more interesting sounds.

OK, we'll cut the lecture off here today and get into actually making a patch from scratch.

Sound fundamentals (part 2)

Last time we got into some of the basics of sound, discussing the two main attributes:

Frequency and Amplitude.

Today we're going to get a little deeper into this stuff so that we can understand how it relates to the music we make using tools like Reason.

OK. So, frequency is basically dealing with pitch, right? How high or low a sound is. Measured in Hertz (Hz). Let's listen to a few tones to recalibrate our ears:

So what we were listening to were pure tones. Here is a picture of what a pure 440 Hz tone (aka "Concert A") looks like:
What we have been looking at and listening to so far are what are called sine waves. Sine waves are basically pure tones - nice and smooth and even, no additives or preservatives. But, as with so many things in life, reality is almost never so smooth and even...

Here is what Concert A (aka 44oHz) looks like when played by a grand piano:

Whoa. Lots of stuff going on here. I should point out that this picture is zoomed way out, so we're not seeing the individual waves like we were in the picture of the sine wave. But my point is this: there is a LOT more than just a simple 440Hz sine wave playing when you hit the A key on a piano.

Major point #1: Almost no natural sound contains only one frequency.

You might be asking yourselves then, "What are the other frequencies?"

Short answer: harmonics.

Harmonics are whole number multiples of a specific frequency.

OK, so I just lost about half the class with that last sentence. But it's really not that complicated. Check it out:


So, we're looking at the first 5 harmonics of a vibrating string. The first harmonic is what is called the fundamental frequency. The fundamental is like the "main note" being played. For example, in the picture of the piano note above, 440 Hz is the fundamental, but all that other stuff in the waveform is a bunch of harmonics:

1st harmonic (aka fundamental) = 440Hz
2nd harmonic = 880Hz (440 x 2)
3rd harmonic = 1320 Hz (440 x 3)
4th harmonic = 1760 Hz (440 x 4)
5th harmonic = 2200 Hz (440 x 5)

Every musical instrument has harmonics, but the amounts and combinations of these harmonics are unique to every instrument. This is why a guitar sounds like a guitar, a snare drum like a snare drum, Mariah Carey like Mariah Carey, etc.

Here is a comparison of a flute, a clarinet, an oboe, and a saxophone all playing Middle C (256 Hz):

As you can see, they are similar (they are all instruments from the woodwind family), but it is the unique harmonic content that gives each one a unique sound.

Major point #2: an octave is a doubling of frequency.

Remember our old friend the octave? A couple classes ago I described an octave as "a set of all the notes you can possibly play".

This is true. That is a musical way of understanding what an octave is.

But you can also look at it from a scientific perspective and see that everytime you go up an octave, you are doubling the fundamental frequency of the key that you started from.

Now that you know about octaves, see if you can answer this question:
Even-numbered harmonics are generally considered to be more musically pleasing than odd numbered ones. Why do you think this might be?

1st harmonic (aka fundamental) = 440Hz
2nd harmonic = 880Hz
3rd harmonic = 1320 Hz
4th harmonic = 1760 Hz
5th harmonic = 2200 Hz


Major point #3: Using electronic equipment (aka synthesizers), it is possible for humans to create all kinds of original sounds that don't exist in nature.

We will be getting deeper into this later when we discuss synthesizers, but for now let's just be aware that there are a handful of common types of sound waves that are used as building blocks in creating electronic sounds and instruments. They are:


1. Sine wave - Our old friend. The simplest of all sounds. A pure frequency.
2. Square wave - A tone with an infinite set of odd harmonics. Pretty harsh sounding, but it's a good material to start with when you're creating new sounds.
3. Triangle wave - A tone that also has an infinite set of odd harmonics, but they fall off in volume more quickly than the square wave.
4. Sawtooth wave - A tone that has all the related harmonics.


We'll stop here for the moment. In the next lesson we will learn about phase, filters and volume envelopes.

Oh yes.