Preparation and Process

Last month we touched on the digital process.

This month we are going to talk about the preparation, the signal path, dos and don’ts and what some of the terminologies mean.

The most important part of the sampling process is preparation. If you prepare properly, then the whole sampling experience is more enjoyable and will yield you the optimum results.
Throughout this tutorial, I will try to incorporate as many sampler technologies as possible, and also present this tutorial side by side, using both hardware and software samplers.

So let us start with the signal path. Signal, being the audio you are recording and path, being the route it takes from the source to the destination.

The signal path is the path that the audio takes from it’s source, be it a turntable, a synthesizer etc, to its final destination, the computer or the hardware sampler. Nothing is more important than this path and the signal itself. The following list is a list of guidelines. Although it is a general guide, it is not scripture. We all know that the fun of sampling is actually in the breaking of the so-called rules and coming up with innovative ways and results. However, the guide is important as it gives you an idea of what can cause a sample to be less than satisfactory when recorded. I will list some pointers and go into more detail about each pointer.

  • The more devices you have in the signal path, the more the sample is degraded and coloured. The more devices in the path, the more noise is introduced into the path, and the headroom is compromised depending on what devices are in the path.
  • You must strive to obtain the best possible S/N (signal to noise ratio), throughout the signal path, maintaining a hot and clean signal.
  • You must decide whether to sample in mono or stereo.
  • You must decide what bit depth and sample rate you want to sample at.
  • You need to understand the limitations of both the source and destination.
  • You need to understand how to set up your sampler (destination) or sound card (destination) to obtain the best results.
  • You need to understand what it is that you are sampling (source) and how to prepare the source for the best sampling result.
  • If you have to introduce another device into the path, say a compressor, then you must understand what effect this device will have on the signal you are sampling.
  • You must understand what is the best way to connect the source and destination together, what cables are needed and why.
  • You need to calibrate the source and destination, and any devices in the path, to obtain the same gain readout throughout the path.
  • You need to understand the tools you have in the destination.
  • Use headphones for clarity of detail.

Basically, the whole process of sampling is about getting the audio from the source to the destination, keeping the audio signal strong and clean, and being able to listen to the audio in detail so you can pick out any noise or other artifacts in the signal.

In most cases, you can record directly from the source to the destination without having to use another device in the path. Some soundcards have preamps built into their inputs, along with line inputs, so that you can directly connect to these from the source. Hardware samplers usually have line inputs, so you would need a dedicated preamp to use with your microphone, to get your signal into the sampler. The same is true for turntables. Most turntables need an amp to boost the signal. In this instance, you simply use the output from the amp into your sampler or soundcard (assuming the soundcard has no preamp input). Synthesizers can be directly connected, via their outputs, to the inputs of the hardware sampler, or the line inputs of the soundcard.

As pointed out above, try to minimise the use of another device in the path. The reason is quite simple. Most hardware devices have an element of noise, particularly those that have built-in amps or power supplies. Introducing these in the signal path adds noise to the signal. So, the fewer devices in the path, the less noise you have. There are, as always, exceptions to the rule. For some of my products, I have re-sampled my samples through some of my vintage compressors. And I have done it for exactly the reasons I just gave as to why you must try to not do this. Confused? Don’t be. I am using the character of the compressors to add to the sample character. If noise is part of the compressor’s character, then I will record that as well. That way, people who want that particular sound, influenced by the compressor, will get exactly that. I have, however, come across people who sample with a compressor in the path just so they can have as strong and pumping signal as possible. This is not advised. You should sample the audio with as much dynamic range as possible. You need to keep the signal hot, ie as strong and as loud as possible without clipping the soundcard’s input meters or distorting in the case of hardware samplers. Generally, I always sample at a level 2 dBu below the maximum input level of the sampler or soundcard, ie 2 dBu below 0. This allows for enough headroom should I choose to then apply dynamics to the sample, as in compression etc. Part 1 of these tutorials explains dynamic range and dBs, so I expect you to know this. I am a vicious tutor, aren’t I? He, he.

My set up is quite simple and one that most sampling enthusiasts use.

I have all my sources routed through to a decent quality mixer, then to the sampler or my computer’s soundcard. This gives me great routing control, many ways to sample and, most important of all, I can control the signal better with a mixer. The huge bonus of using a mixer in the path and as the heart of the sampling path is that I can apply equalisation (eq) to the same source sample and record multi takes of the same sample, but with different eq settings. This way, by using the same sample, I get masses of variety. The other advantage of using is a mixer is that you can insert an effect or dynamic into the path and have more control over the signal, than just plugging the source into an effect unit or a compressor.

Headphones are a must when sampling. If you use your monitors (speakers) for referencing, when you are sampling, then a great deal of the frequencies get absorbed into the environment. So, it is always hard to hear the lower noise or higher noise frequencies, as they get absorbed by the environment. Using headphones, either on the soundcard, or the sampler, you only hear the signal and not the environment’s representation of the signal. This makes finding noise or other artifacts much easier.

The decision of sampling in mono or stereo is governed by a number of factors, the primary one being that of memory. All hardware samplers have memory restrictions, the amount of memory being governed by the make and model of the sampler. Computer sampling is another story entirely, as you are only restricted by how much ram you have in your computer. A general rule of thumb is: one minute of 44.1 kHz (audio bandwidth of 20 kHz using Nyquist theorem, which I covered in Part 1) sample rate audio, in stereo, equates to about 10 megabytes of memory. Sampling the same sampling rate audio in mono gives you double the time, ie 2 minutes, or takes up 5 megabytes of memory.

So, depending on your sampler’s memory restriction, always bear that in mind. Another factor that governs the use of mono over stereo is, whether you actually need to sample that particular sound in stereo. The only time you sample in stereo is if there is an added sonic advantage in sampling in stereo, particularly if a sound sounds fuller and has varying sonic qualities, that are on the left and right sides, of the stereo field, and you need to capture both sides of the stereo field. When using microphones on certain sounds, like strings, it is often best to sample in stereo. You might be using 3 or 4 microphones to record the strings, but then route these through your mixer’s stereo outputs or subgroups to your sampler or soundcard. In this case, stereo sampling will capture the whole tonal and dynamic range of the strings. For those that are on stringent memory samplers, sample in mono and, if you can tolerate it, a lower sampling rate. But make sure that the audio is not compromised.

At this point, it is important to always look at what it is that you are sampling and whether you are using microphones or direct sampling, using the outputs of a device to the inputs of the sampler or soundcard. For sounds like drum hits, or any sound that is short and not based on any key or pitch, like instrument or synthesizer sounds, keep it simple and clean. But what happens when you want to sample a sound from a particular synthesizer? This is where the sampler needs to be set up properly, and where the synthesizer has to be set up to deliver the best possible signal, that is not only clean and strong but one that can be easily looped and placed on a key and then spanned. In this case, where we are trying to sample and create a whole instrument, we need to look at multi-sampling and looping.

But before we do that, we need to understand the nature of what we are sampling and the tonal qualities of the sound we are sampling. Invariably, most synthesizer sounds will have a huge amount of dynamics programmed into the sound. Modulation, panning, oscillator detunes etc are all in the sound that you are trying to sample. In the case of analog synthesizers, it becomes even harder to sample a sound, as there is so much movement and tonal variances, that it makes sampling a nightmare. So, what do we do? Well, we strip away all these dynamics so that we are left with the original sound, uncoloured through programming. In the case of analog synthesizers, we will often sample each and every oscillator and filter. By doing this, we make the sampling process a lot easier and accurate. Remember that we can always program the final sampled instrument to sound like the original instrument. By taking away all the dynamics, we are left with simpler constant waveforms, that are easier to sample and, more importantly, easier to loop.

The other consideration is one of pitch/frequency. To sample one note is okay, but to then try to create a 5 octave preset presentation of this one sample would be a nightmare, even after looping the sample perfectly. There comes a point that a looped sample will begin to fall out of pitch and result in a terrible sound, full of artifacts and out of key frequencies. For each octave, the frequency is doubled. A way around this problem is multi-sampling. This means we sample more than one note of the sound, usually each third or fifth semitone. By sampling a collection of these notes, we can then have a much better chance of recreating the original sound accurately. We then place these samples in their respective ‘slots’ in the instrument patch of the sampler or software sampler, so a C3 note sampled, would be put into a C3 slot on the instrument keyboard layout. Remember, we do not need to sample each and every note, just a few, that way we can span the samples, ie we can use a C3 sample and know that it can still be accurate from a few semitones down to a few semitones up, so we spread that one sample down a few semitones and up a few semitones. These spread or zones are called keygroups. Emu call these zones and Akai call them keygroups. Where the sample ends, we put our next sample and so on, until the keyboard layout is complete with all the samples, this saves us a lot of hard work, in that we don’t have to sample every single note, but also gives us a more accurate representation of the sound being sampled. However, multi-sampling takes up memory. It is a compromise between memory and accurate representation that you need to decide on.

There are further advantages to multi-sampling, but we will come to those later. For sounds that are more detailed or complex in their characteristics, the more samples are required. In the case of a piano, it is not uncommon to sample every second or third semitone and also to sample the same notes with varying velocities, so we can emulate the playing velocities of the piano. We will sample hard, mid and soft velocities of the same note and then layer these and apply all sorts of dynamic tools to try to capture the original character of the piano being played. As I said, we will come to this later.

An area that is crucial is that of calibrating. You want to make sure that the sound you are trying to sample has the same level, as shown on the mixer’s meters, as the sampler’s meters or the soundcard’s meters. If there is a mixer in the path, then you can easily use the gain trims on the mixer, where the source is connected to, to match the level of the sound you want to sample, to the readout of the input meters of the sampler or the soundcard. If there is no mixer in the path, then you need to have your source sound at maximum, assuming there is no distortion or clipping, and your sampler’s or soundcard’s input gain at just below 0dBu. This is a good hot signal. If you had it the other way around, whereby the sound source level was too low and you had to raise the gain input of the sampler or soundcard, you would then be raising the noise floor. This would result in a signal with noise.

The right cabling is also crucial. If your sampler line inputs are balanced, then use balanced cables, don’t use phono cables with jack converters. Try to keep a reasonable distance between the source and destination and if you have an environment with RF interference, caused by amps, radios, antennae etc, then use shielded cables. I am not saying use expensive brands, just use cables correctly matched.

Finally, we are left with the tools that you have in your sampler and software sampler.

In the virtual domain, you have far more choice, in terms of audio processing and editing tools, and they are far cheaper than their hardware counterparts. So, sampling into your computer will afford you many more audio editing tools and options. In the hardware sampler, the tools are predefined.

In the next section, we will look at some of the most common tools used in sampling.

Additional content:

Preparing and Optimising Audio for Mixing

Normalisation – What it is and how to use it

Topping and Tailing Ripped Beats – Truncating and Normalising

Most sampling enthusiasts usually sample a beat, audio piece or riff when they sample. Your sampler is so much more than that, and offers a wealth of tools that you rarely even knew existed, as they are kept so quiet, away from the ‘in your face’ tools.

This tutorial aims to open your eyes to what you can actually achieve with a sampler, and how to utilise what you sample.

This final tutorial is the real fun finale. I will be nudging you to sample everything you can and try to show you what you can then do to the sample to make it usable in your music.

First off, let us look at the method.

Most people have a nightmare when it comes to multi-sampling. The one obstacle everyone seems to be faced with is how to attain the exact volume, length of note (duration) and how many notes to sample.

The easy method to solve these questions in one hit is to create a sequence template in your sequencer. This entails having a series of notes drawn into the piano roll or grid edit of your sequencer. You can actually assign each and every note to be played at a velocity of 127 (maximum volume), have each note the exact same length (duration) and you can have the sequencer play each and every note or any number of notes you want. The beauty of this method is that you will always be triggering samples that are at the same level and duration. This makes the task of looping and sample placing much easier. You can save this sequence and call it up every time you want to sample.

Of course, this only works if you have a sequencer and if you are multi-sampling. For sampling the source directly, as in the case of a synth keyboard, it is extremely useful.

Creative Sampling

The first weapon in creative sampling is the ‘change pitch’ tool. Changing the pitch of a sample is not just about slowing down a Drum and Bass loop until it becomes a Hip Hop loop, a little tip there that some people are unaware of. It is about taking a normal sound, sampling it then pitching it right down, or up, to achieve a specific effect.

Let us take a little trip down the ‘pitch lane’.

You can achieve the pitch down effect by using the change pitch tool in your sampler, assigning the sample to C4 then using the C1 note as the pitched-down note, or time stretch/compress to maintain the pitch but slow or speed the sample. There is a crucial distinction here. Slowing down a sample has a dramatic effect on the pitch and works great for slowing fast tempo beats down to achieve a slower beat, but there comes a point where the audio quality starts to suffer and you have to be aware of this when slowing a sample down. The same is true for speeding a sample up. Speed up a vocal sample and you end up with squeaky vocals.

Time stretching/compressing is a function that allows the length of a sample to be changed without affecting the original pitch. This is great for vocals. Vocals sung for a track at 90 BPM can then be used in a track at 120 BPM without having to change the pitch. Of course, this function is as good as the software or hardware driving it. The better the stretching/compressing software/hardware is, the better the result. Too much of stretching/compressing can lead to side effects, and in some cases, that is exactly what is required. A flanging type of robotic effect can be achieved with extreme stretching/compressing, very funky.

A crucial function to bear in mind, and always perform, is that when you pitch a sample down, you then need to adjust the sample start time. Actually, this is a secret weapon that programmers and sound designers use to find the exact start point of a sample. They pitch the sample right down and this makes it much easier to locate the start point. You will often find that a sample pitched down a lot will need to have the start time cropped, as there will be dead air present. This is normal, so don’t let it worry. Simply check your sample start times every time you perform a pitch down.

Here are a few funky things to sample.

Crunching, flicking, hitting paper

Slowly crunch a piece of paper, preferably a thicker crispier type of paper, and then sample it. Once you have sampled it, slow it right down and listen to the sample. It will sound like thunderclaps. If you are really clever you can listen to the sample as you slow it down, in stages, until you hear what sounds like a scratch effect, before it starts to sound like thunderclaps. SCSI dump the samples into your computer, use Recycle or similar, and dump the end result back into your sampler as chopped segments of the original sample (please read ‘chopping samples’ and ‘Recycle tutorial’).

Big sheets of paper being shaken or flicked from behind can be turned into thunderous noises by pitching down, turning up and routing through big reverbs.

Spoon on glass

There are two funky ways to do this. The first is with the glass empty. Use an empty glass, preferably a wine glass, and gently hit it with a spoon. Hit different areas of the glass as this generates different tones. You can then slow these samples down till you have bell sounds, or keep them as they are and add reverb and eq to give tine type of sounds.

The second way of doing this is to add water to the glass. This will deaden the sound and the sample will sound a lot more percussive. These samples make for great effects.

Lighting a match

Very cool. Light a match, sample it and slow it down. You will get a burst effect or, being clever, use the attack of the match being lit sample and you will get a great snare sound, dirty and snappy.

Tennis ball against wood

Man, this is a very cool one. Pitch these samples down for kick and tom effects. You can get some really heavy kicks out of this sample. Actually, the ball hitting woody type of surfaces make for great percussive sounds.

Finger clicking

Trim the tail off the sample and use the attack and body of the sample. You now have a stick or snare sound. Pitch it down and you will have a deep tom burst type of effect. Or, use the sample of the finger click, cut it into two segments, the first being the attack and the body, the second being the tail end. Layer them together and you have a snare with a reverse type of effect.

Hitting a radiator with a knife

Great for percussive sounds. Pitched down, you get percussive bells, as opposed to bells with long sustain and releases. Also, if you only take the attack of this sample, you will have a great snare sound.

Kitchen utensil

These are the foundation for your industrial sounds. Use everything. First, drop them all on a hard surface, together. Sample that and slow it down a bit and you will have factory types of sounds. Second, drop each utensil on a hard surface and sample them individually. They make for great bell and percussive sounds. Scrape them together and sample them. Slowed down, they will give you great eerie industrial sounds and film sound effects. Metallic sounds, once pitched down, give more interesting undertones, so experiment.

Hitting a mattress with a piece of wood

This will give a deep muffled sound that has a strong attack. This makes for a great kick or snare. Slowed right down, you will achieve the Trancey type of deep kick.

Blowing into bottles

This gives a nice flute type of sound. Pitched down, you will get a type of foghorn sound. Blow into it short and hard and use the attack and body, you will achieve a crazy deep effect when pitched down.

Slamming doors

Slam away and sample. Thunderous sounds when pitched down. The attacks of the samples make for some great kicks and snares.

Aerosol cans

Great for wind and hi-hats. Slowed down, you will achieve wind type sounds. Used as pitched up, you get cabasa type of sounds. Run through an effect and pitched higher, you will achieve a hi-hat type of sound.

Golf ball being thrown at a wall

A snare sample that is great in every respect. Kept as is, you get a cool snare. Pitched up and you get a snappier snare. Pitched down, you get a deep tom, kick or ethnic drum sound.


Sample toys, preferably the mechanical and robotic ones. The number of sample variations you will get will be staggering. These mechanical samples once pitched down, make for great industrial sounds. Pitched up, they can make some great Star Wars type of sounds. Simply chopped up as they are, make for great hits, slams and so on.

Factories and railway stations

Take your recorder and sample these types of locations. It is quite amazing what you will find and once manipulated, the samples can be so inspiring.

Toilets, sinks, and bathtubs.

Such fun. Water coming out of a tap pitched down can be white water. Water dripping can be used in so many ways. Splashing sounds can be amazing when pitched up or down. Dropping the soap in a full bath and hitting the sidewalls of the bathtub when empty or even full, can create some of the best percussive sounds imaginable.


Sample your radio, assuming it has a dial. The sounds of searching for stations can give you an arsenal of crazy sounds. Pitched down you will get factory drones, swirling electric effects and weird electro tom sounds. The sound palette is endless.

I think you get the picture by now. Sample everything and store it away. Create a library of your samples. Categorise them, so that they are easy to locate in the future.

Now let us look at what you can do to samples to make them interesting.

Reverse is the most obvious and potent tool. Take a piece of conversation between a man and a woman, sample it and reverse it and, hey presto, you have the Exorcist.

Layer a drum loop with the reversed version of the loop and check it out. Cool.

Pitch the reversed segment down a semitone or two to create a pseudo doppler effect.

With stereo samples of ambient or melodic sounds, try reversing one channel for a more unusual stereo image. You can also play around with panning here, alternating and cross-fading one for the other.

Try sampling at the lowest bandwidth your sampler offers for that crunchy, filthy loop. This is lo-fi land. Saves you buying an SP1200..he..he.

Try deliberately sampling at too low a level, then using the normalising function repeatedly to pump the volume back up again. This will add so much noise and rubbish to your sample that it will become dirty in a funky way.

You can take a drum loop and normalise it continually till it clips heavily. Now Recycle the segments, dump them back into your sampler, and you have dirty, filthy, crispy Hip Hop cuts.

A sample doubles its speed when it’s transposed up an octave. So try triggering two versions of a sampled loop an octave apart, at the same time. With a percussive loop, you’ll get a percussion loop running over the top of the original.

Use effects on a loop, record it to cassette for that hissy flavour, then, resample it. Recycle the whole lot and drop the segments back into your sampler and you have instant effects that you can play in any order.

Layer and cross-fade pad samples so that one evolves/morphs into another.

Take a loop and reverse it. Add the reversed loop at the end of the original loop for some weirdness.

Multi triggering a loop at close intervals will give you a chorus or flange type of effect. Try it. Have the same loop on 3 notes of your keyboard and hit each note a split second after the other. There you go.

I could go on for pages but will leave you to explore and enjoy the endless possibilities of sampling and sound design.

Additional content:

Preparing and Optimising Audio for Mixing

Normalisation – What it is and how to use it

Topping and Tailing Ripped Beats – Truncating and Normalising

Most turntables that are stand-alone will require a preamp to boost the signal so that you can record an acceptable level. Some turntables, particularly those that are housed in hi-fi units, will have an amp built-in, but for the more pro decks, or DJ turntables, a preamp is required. The choice of the preamp is crucial. I could go into some very deep explanation about capacitance, hum, LF noise and impedance etc but that would confuse you at this stage. What I will say is that the following will save you great heartache and make life a great deal easier.

Years back, the RIAA (Recording Industry Association of America) established what is known as compensation standards. The resulting RIAA preamp has been built into every hi-fi and stereo amp with phono or turntable inputs since then. In the event that you are using a turntable, connected to a mixer or stand-alone, that does not have a built-in RIAA preamp, then you would need to get one. Now, this is where the technical heads sometimes have a fiery debate. Do you apply RIAA equalisation at the preamp stage or after using software applications? Take my word for it, always apply the RIAA equalisation at the analog stage, at the preamp, and not later. This will ensure a good strong dynamic signal with ample headroom.

Most vinyl is made with what we call pre-emphasis, a type of EQ, to tame the amount of low-frequency energy recorded on the disc. This pre-emphasis then has to be corrected using what’s known as an RIAA curve, which boosts the low end and reduces the high end so that an overall flat frequency response is shown.

Additionally, the output signal then needs amplification and RIAA correction.

In today’s world of sampling genres, it is crucial to have an industry-standard approved amp to power your turntable signal. Use only RIAA approved amps, or similar in specification, as the integrity and quality of the signal you are amplifying are as important as the signal chain it goes through.

Relevant content:

Sampling Tools and Procedures

Preparing to Sample – Using hardware samplers!

Sampling Techniques and Best Practices

This month’s tutorial is going to concentrate on the basic and general tools available for the sampling process and will not focus on the more detailed or esoteric tools that are adopted to further hone the sample.

So, let’s start right at the input stage of the sampler or sound card.

We have already covered the topic of attaining a clean and hot signal. Now, we need to cover the tools available to actually sample a sound, and how to use the tools available after you have sampled a sound.

Most samplers will allow you to sample in a number of ways. But first, it is important, and sensible, to create a location for the samples. On computers, it is always good practice to create a section on your hard drive for audio. You can then create folders for your samples and have them in categories, for example, if you are sampling bass sounds, have a folder named ‘Basses’, for drums have a category named ‘Drums’ and then assign subcategories and name them relative to what you are sampling. So, for Drums, you c could have subcategories for kicks, snares, and hi-hats etc. This makes filing (archiving) of the samples, and even more importantly, the searching for a sample, much easier.

On hardware samplers, it is pretty much the same. You create a bank and name that and within that bank you create presets, which house the samples. On Emu samplers, the sampler creates a default preset on startup. This makes life easier. Most samplers have this facility.

Now let us look at the different ways of sampling that certain samplers provide.

I am going to concentrate on Emu Ultra samplers for this tutorial.

For the sampler to begin sampling, it needs to know a few things.

  • Source analog or digital, 44.1 kHz or more. Pretty self-explanatory as it is asking you to choose the source and the sample rate. Some samplers will have the option that will allow for digital recording as well as analog. There are advantages to using digital recording modes but there are also disadvantages. The sample rate, however, is important. If you own a Sound Blaster card and it only operates at 48 kHz, then sampling at 44.1 kHz is not helpful at all. The other advantage of sampling at a higher rate is for precision and clarity in the representation of the sound you are sampling. The disadvantage of higher sample rates is that they will eat up memory. In the virtual world (computer), it is now more common to sample at 24 Bits and 96 kHz (24/96) or 44.1 kHz (24/44.1). However, these parameters are dependent on the sound card you are using. If 24/96 is not supported then you cannot sample at those values.
  • Input This is for selecting mono or stereo for the sampling process.
  • Length You can predetermine the length of the sample you want to record. Maybe you only need to sample 3 seconds of a sound. Setting 3 seconds as the length automatically stops the sampler recording after 3 seconds of sampling.
  • Dither Used when recording digitally.
  • Monitor Gives you the option of having it on or off. Setting it to on allows you to listen to the sound being sampled while it is being sampled.
  • Gain Here you can adjust the input gain (volume/level) of the sound (signal) being sampled. If the signal is too loud and is distorting or clipping, you can adjust the level by using this function.
  • Trigger Key This is one of the methods of sampling that I mentioned earlier. You can set the trigger key to any key on the keyboard, say C4, and when you hit C4 on your keyboard the sampler activates (gets triggered) and starts to sample.
  • Arm This puts the sampler into standby mode and when it hears a signal, it starts to sample. This is usually used in conjunction with the threshold. The threshold sets the level at which you want the sampler to be triggered when in arm mode. The real advantage of this is to eliminate noise. If you set the threshold above the noise level and then play the sound, the sampler will only start to record at the threshold level setting, in this case, above the noise, as the noise is below the threshold level. The threshold/arm combination is also useful when you want to sample a sound that is above the general level of the piece of audio being sampled, an example of this would be to sample a loud snare that is above the rest of the audio piece. If you set the threshold to just below the level of the snare, the sampler ignores everything below that level and automatically records the snare.
  • Force or Manual This simply means that you press a button to start the sampler recording.

Those are the general functions available on most samplers, to do the actual sampling/recording. Now we need to look at the tools available when you press stop or complete the recording of the sound.

When you press stop, a new page appears and you are given a bunch of options. Here are the general options that are offered to you.

  • Dispose or Keep This just means you can either dump the sample, if it was no good, or keep it.
  • Place This allows you to place the sample anywhere on the keyboard you want and within this option, you will have a range you can set. The range is displayed as Low and High. Let’s say I sample a C3 bass note off a synthesizer, I can then place it at C3 on my keyboard and set the low to A2 and the high to D#3. I have not placed the note and set it a range on my keyboard. This saves me loads of time and effort in having to do it later. This placing and range setting are stored in the preset, so, in effect, I am creating and building my preset as I am sampling, instead of having to sample all the notes then go back into the preset and start placing and setting ranges. Much easier. With a drum loop, you can do the same thing and by setting the range, it gives you different pitch choices of the drum loop as the sample pitches down when setting the low range value, and pitches up when setting the high range value. For single drum shots, I would place and set the ranges at the placed note. So, a kick would be placed on C1 and the low range value would be set to C1 and the same for the high range value. I now have a kick on C1.
  • Truncate Some samplers have auto-truncate and manual truncate. Truncate, also called trim or crop, is a function used to cut data before and after the sampled data. This can cut/delete space or sound before or after the sample or can be used to cut/delete any portion of the sample. Auto-truncate simply removes everything before and after the sample.
  • NormalizeorNormalise This is a topic that has ensured some fiery debates and I doubt it will ever get resolved. Basically normalising a sample means that you raise the volume of the sample to the peak of the headroom. If, for example, you were normalising a sample to 0dB, then that means the process takes the highest peak/s in the sample data and raises them to 0dB, in this way, the loudest peak hits 0dB. This is called Peak (or absolute) normalising. By raising the highest peak you also raise the entire sample data, this has the disadvantage of raising the noise floor as well, as all data is raised till the peak hits 0dB. To normalise an audio file to ensure a certain level of perceived loudness, you need to normalise to an RMS (or relative) value of dB, rather than peak. RMS is, roughly, the average volume over a given time, rather than just the highest peak/s. It calculates the average peaks and raises those to 0dB. The disadvantage of RMS normalising is that by raising the average data peaks, you incur clipping, not always, but usually. So, in this instance, a good normalising plugin will compress or limit at the same time as normalising, so the levels do not exceed 0dB and thus, prevent clipping. I have always maintained that if you have a strong signal, with good dynamic movement, that does not clip and stays just under 0dB, then you are far better off than normalising to 0dB. Of course, there are instances where normalisation can be your friend, but in most cases, it can cause additional side effects that are not needed. These include killing any headroom that was there, raising the noise floor so noise is also now more pronounced and evident, and to top it all off, you can get roundness in the shape of the peaks and even slight distortion or phasing. So, use this function sensibly.

Now you have your sample recorded, placed, truncated, normalised etc, you need to look at the tools available to edit and process the sample.

By selecting ‘edit sample’ you are presented with the sample and a host of tools you can use to edit and process the sample. Let us look at these briefly and then, when we come to the bigger topics, we will get a little more in-depth.

  • Zoom +- This is like a magnifying tool that allows you to zoom in, or magnify, a portion of a sample.
  • Start End Size Here, you have the start and end of the sample represented in cycles and, in some samplers, in time. The size tells you the size of the sample. This might not seem important now but the size of the sample is important when working out and retuning the sample or changing the sample rate. Don’t worry, we will tackle that later.
  • Loop This is a crucial function and is the essence of what a sampler really does. The whole concept of looping is actually a simple one, whether it’s for memory saving or for creating sustained instrument sounds, the process is invaluable. What is difficult is how to find good loop points, and there are a number of reasons why this can seem complex. Firstly, unless the shape of the sample at the beginning, during and end of the loop matches up in level, shape, and phase, you will have problems in finding a clean loop point. The most common enemy here is click. The best way to avoid clicks is to find what we call the ‘zero-crossing’ point. This is where the sample’s shape crosses over from the positive axis to the negative axis. At the point where the shape crosses the axis, we have a zero point. Looping at zero points eradicates the problem of clicks. But, if the shape and level don’t match up well, you will still get a click. So we are still left with a problem. What does this tell us? It tells us that the sample length being looped must be consistent, both in terms of shape and level, but also in terms of length. Too long a sample loop length and you encounter modulation. Why? Because the sample has an attack and decay. If you start your loop point too close to the attack and your endpoint too near the decay, you are then left with a shape that starts high and drops to a lower level, this causes the loop to modulate or wobble up and down. The opposite is also true. All sounds have a harmonic structure and if your loop length is too small then the harmonics of the sound are compromised since you are looping a very small instance of the sample, you are, in effect, cutting the harmonics up. This will give you an unnatural loop in that it will sound very synthesized. That’s ok if you are sampling synthetic sounds but not if you are trying to loop a natural instrument sound. The final problem you are faced with is pitch. If you loop the wrong are of the sample, then it might not be in the right pitch of the original signal that was being sampled. A C3 string note will not stay at exactly C3 but move through the harmonics, so if you looped the wrong harmonic, the sample might show up as C3 +3 or worse, ie it is 3 cents off the right pitch. You need to select the most consistent part of the sample to attain the right loop points and loop length. This, unfortunately, takes practice and experience. This leads me subtly to the next function.
  • Auto Correlation Some samplers provide this function when you are looping. Basically what this function does is, after you have set your loop points, it searches for the next best loop point that it thinks will give you the best loop. Not always accurate but useful to use if you are completely off target. However, we do have another weapon at our disposal if the loop still throws up a click.
  • Crossfade Looping This technique involves fading out the end of the loop and overlapping it with a fade-in of the start of the loop, and it’s a facility provided by virtually all samplers. By fading in these points, you get a smoother transition on the loop points, start and end. I only recommend using this when you have got really close to finding the right loop point and length, as it is a nice little tool and is just a polisher and not a tool to remedy bad loop points and lengths. If you had a very bad loop and it was glitching heavily, then using this tool would only make the sound unnaturally modulated, without any consistent shape. So, it’s not for error correction but for polishing off the tiny click that might be barely audible.
  • DC offset Any waveform that isn’t symmetrical around the zero axis has a DC offset. DC offset is when there is too large a DC (direct current) component in the signal, sometimes visible as the signal not being visually ‘centered’ around the ‘zero level axis’. DC offsets do not affect what you actually hear, but they affect zero-crossing detection and certain processing, and it is recommended that you remove them. That’s the technical, but short, definition. Basically, always remove the DC offset on a sample. This will help you find zero-point crossings. This is a whole debate in itself and there are arguments raging on both sides of the fence and arguments based around the algorithms used in DC offsetting tasks. You don’t need to even think about getting involved in this debate. What you do need to know and do is to remove the DC offset on a sample and you are usually given a tool in the menu option to do this. The DC offset removal is actually called the DC filter, for those who want to know. Try experimenting, as always.

Relevant content:

RIAA Amps and Standards

Preparing to Sample – Using hardware samplers!

Preparing and Optimising Audio for Mixing

Normalisation – What it is and how to use it

Topping and Tailing Ripped Beats – Truncating and Normalising

I find that the most common hurdles that beginners face are that of understanding how to use their samplers, how to hook all the devices up to each other, and how to then manage the samples. The best way of tackling these sub-topics is to give you some pointers and guides, and from there, you should be able to perform the task of sampling in a coherent and ordered fashion.

Sampling is not about just recording a piece of audio, it is about organisation, management and following a protocol that ensures the best results. If these criteria are not adhered to, then you will always struggle and, more often than not, be totally disheartened by the process and results. Practice is the answer, but to be effective, one needs to follow procedure, otherwise bad habits will develop and breaking those habits becomes harder and harder with time.

Whether you are sampling in a hardware environment or software environment, the methodology is the same. You need to have a temporary location for your samples, for editing and processing, and a final destination for the samples you want to keep. For this, we have to create directories. Within those directories, we need to create sub-directories. This ensures a simple way of locating samples and makes for a neater and logical layout. So, in the case of soft sampling, ie in a computer, we need to create folders with sensible names. In the case of percussion, it makes sense to name the main folder ‘Drums’. We can then create sub-folders within the main folder and name those, for example, we could create folders with names like ‘Kicks’, ‘Snares’, ‘Hi-Hats’ and so on. We can then create another main folder and name that ‘Loops’. We can then create sub-folders and name those in accordance to BPM(Beats per minute) or genre-specific or both. An example would be ‘Hip Hop’, sub-folder ’60-85 BPM’ etc…This makes life so much easier. We can continue this method and create more folders for instrument samples or loops. You get the picture? Organisation is crucial and order is paramount. The same applies to hardware samplers. There exists, in all hardware samplers, naming and filing options. This method of archiving should be done prior to any sampling to ensure that you have a trouble-free way of following the process and retrieving the data at any time.

We now come to the path. As discussed in earlier parts of this tutorial, the signal path is the most important aspect of sampling. Keeping the signal clean and strong minimises the noise element and ensures the best dynamic range. But this is always the area that beginners struggle with. The reason for this is the lack of understanding of gain structures and the devices in the chain. Let me make that simpler to understand. Most beginners make rudimentary errors when sampling because they do not understand the nature of the sound they are sampling or the equipment being used in the signal path. The most common errors are that of recording a distorted signal, due to too high a gain, recording too low a signal, which results in adding noise when the sample is then normalised or the gain increased or encountering hum because they had to use a preamp to boost the turntable signal to be able to sample it, or when everything is absolutely right, there is still noise or hum or any artifact that cannot be traced. Of course, there are more errors than that, but these are the most basic and yet the most common, so maybe we should tackle these problems before we continue.

So, to help you understand and set up your devices a bit better the following hints and definitions will hopefully help you a tad.

1. Using a turntable

Most turntables that are stand-alone will require a preamp to boost the signal so that you can record an acceptable level. Some turntables, particularly those that are housed in hi-fi units, will have an amp built-in, but for the more pro decks, or DJ turntables, a preamp is required. The choice of a preamp is crucial. I could go into some very deep explanation about capacitance, hum, LF noise and impedance etc but that would ruin our friendship. What I will say is that the following will save you great heartache and make life a great deal easier.

Years back, the RIAA (Recording Industry Association of America) established what are known as compensation standards. The resulting RIAA preamp has been built into every hi-fi and stereo amp with phono or turntable inputs since then. In the event that you are using a turntable, connected to a mixer or stand-alone, that does not have a built-in RIAA preamp, then you would need to get one. Now, this is where the technical heads sometimes have a fiery debate. Do you apply RIAA equalisation at the preamp stage or after using software applications? Take my word for it, always apply the RIAA equalisation at the analog stage, at the preamp, and not later. This will ensure a good strong dynamic signal with ample headroom.

2. Cables

If I had a penny for every time the question of cables comes up, I would be one rich dude.

There are a few things that are crucial about cables and let us also put to bed the ridiculous analogy of ‘Expensive cables are better than cheaper cables’. This is simply not true, and if you actually took the time to make your own cables from component parts, you would realise how cheap it actually is to make your own quality cables. In fact, I will write a tutorial on this soon, along with how to build your own pop-shield. Both are crucial DIY projects that, would save you money, and are fun.


A balanced line requires three separate conductors, two of which are signal (+ and -) and one shield/earth. You can usually determine these by looking at the connection. They will have 2 black rings and the plugs are referred to as TRS (tip, ring, and shield). Sometimes, and not always correctly, referred to as stereo jacks.


An unbalanced cable runs two connectors, a hot (+) and an earth.

By the way, I am being very simplistic here as there are many variations to balanced, unbalanced, TRS, coax, etc…What is important is that if your equipment is balanced, then use balanced cables throughout the path and vice versa. The advantage of using balanced cables is one of noise reduction. Finally, if connecting balanced outputs/inputs with unbalanced cables, you can end up with signal levels that are 6dB lower than they should be. This is essentially because only half the signal is being transferred. So it always pays to match your cables.

You will find that a lot of cables are unbalanced. Guitar jack cables, speaker cables, and microphone cables being the most common.

Shielded cables can also afford better protection against RF (radio frequency) noise.

Match your cables.

Even better, switch to balanced cables, throughout the path, if possible, that way you reduce noise and cable length does not become such an issue. This has subtly led me onto the debate of length. This is, again, dependant on the type of cable and connectors. Generally, as a rule, you can use unbalanced cables with no worries at all, up to 5 metres. Balanced can go even further, 10 metres. However, these figures are not gospel.

Now we will deal with connectors. This is another area that is rife with preferences and arguments. So, I will sum up both the cable and connectors in one statement. I make my own cables but if I have to buy, then I buy Van Damme, Mogami or equivalent, and for connectors, I always use Neutrik connectors, Cannon and Switchcraft follow. My recommendation is, build your own cables. This saves money and teaches you a thing or two.

3. Ground loops, hums, power surges, and other nasty artifacts 

Without going into too much detail as to what factors cause the above, I would rather propose a solution. You now have a little more insight into why certain cables can filter noise better than others, along with connectors and cable lengths and cable matching. What we now need to look at is how to prevent earth loops and surges and even hums. Most equipment needs to be earthed in some fashion and the very nature of our planet and the national grid system means we will have power surges and spikes in our mains. Add to that mains hum, or equipment hum from non-earthed equipment, and you are confronted with a multitude of problems that can all be resolved with a simple and inexpensive solution.

Nowadays, there are a number of companies that build power surge protectors in terms of mains switches, isolators for maintaining a constant predefined current, power distributors for maintaining and distributing current to a number of devices and UPS systems (uninterruptible power supply) for protection against power-downs, cuts, and outages. Simply put, you want to protect your equipment against power surges, spikes, shutdowns, etc. So, the simplest answer is to buy a power distributor that connects to all your equipment in the way of kettle plugs and sockets, a surge protector in the way of a simple mains switch breaker, found at any shop that sells plugs and the like, and that’s pretty much it.

Emo and Furman make good power distributors and protectors and they are cost-effective. Many companies make UPS systems and they can start at a very cheap bracket and go into a hefty price range, the latter being for serious users like hospitals and the like. A simple UPS system can not only protect your system against power cuts, surges, spikes but also act as a distributor for your equipment, and not break the bank either. Most commonly used when you have a computer running in your studio, and a number of other devices, that rely on a constant feed. This way, if there is a power cut in your area, the UPS will have a battery charge back-up and will continue to function, allowing you to back up your data on a computer instead of having it all wiped out by the power cut.

Personally, I have an Emo power distributor that affords me 12 kettle sockets which connect to the gear that cost me £70, and a surge protector plug set that cost me £8 from my local Maplin. If you have serious mains issues, then seek the correct help and, if possible, have an isolator specifically for your studio. If you require a UPS system, then there are a number of cheap manufacturers on the net, APC being one of the most noted. Make sure to match the power and get a True-Online type. Seek them and be happy.

Bear in mind that your turntable may cause ground hum so some type of grounding is required. With the latest Emu sound cards, notably the 1820M, there is a dedicated turntable input with a ground lug. That, to me, is one serious cost-effective way of having a sound card and a preamp with grounding, all in one unit.

4. The sound card

Probably the most confusing and wrought with obstacles is the subject of sound cards. Which one to buy, how to hook it all up, what connections, how to assign the ins and outs, analog or digital, adapt or optical, what sample rate…?

All the above can be daunting for the beginner, but it can be made easy if you understand a few very basic concepts about what the sound card is and how it functions.

As always, the goal here is to get as hot a signal as possible into the computer without noise or distortion or to compromise the headroom.

Some people like to sample digitally as opposed to analog sampling. Remember that we are in the computer’s domain here and not external hardware sampler territory. This is all about connection, so it makes sense to set your sound card’s inputs to match the incoming signal. If you are using any of the digital inputs, ADAT, SPDIF etc, then you need to select those as your inputs from the sound card’s control panel or software on the computer. If you are using the analog inputs, then you need to select these from your computer. I always recommend a hot signal at source, for example, the turntable’s preamp, after selecting the highest gain value without any distortion, you need to match the input signal by adjusting the sound card’s input gains, either from your sound card’s control panel or physically, by adjusting the trims or knobs on the sound card itself, assuming it has any. Check your meter’s in the software application that you are using to record into. Remember that in the digital domain anything above 0dB is clipping, it is not the same for the analog world, where you have some play or headroom in the signal boost. Try to keep your signal a couple of dB below 0, that way you have left enough headroom should you wish to process the sample. If you have a dead-on 0dB recording, and if you apply compression or any dynamics that boost the gain, the sample will clip. Keep it sensible.

The other area we need to touch on is the operating level.

Most pro gear operates at a nominal +4dBu and often with balanced interfaces. Most consumer or semi-pro gear uses a -10dBV operating level, and often with unbalanced interfaces. But the two levels are not interlinked or dependant. You can have +4 unbalanced or -10 balanced. These levels are measured as dBu (.775V), dBV (1V), so you can see that there is a difference in the referencing. I do not expect you to understand this as of yet, but if you want to delve into it a bit deeper, then read my Synthesis tutorials. However, you might come across certain products that are set to nominal operating levels; in this instance the gain staging is important.

5. Matching levels

It is imperative to understand how to calibrate the signal path for optimum signal to noise ratio (S/N) and to also get a true reading so that your levels show the same legending. Basically, what all this means, is that you need to be able to see the same level readouts on your hardware and software so that you are dealing with a known quantity. It is pointless if you have different gain readouts across your signal path. So, what we need to do here is to calibrate the system. In fact, it is essential to do this anyway, so that when you are mixing or producing, your levels are true. By calibrating your system and showing a true value across the path, you are then in a stronger position to be able to apply dynamics that might be dependant on numerical data as opposed to the ‘ear’ concept, that of hearing.

So, let us start at the source and finish at the destination. In this instance, the source will be the turntable, microphone or synthesizer and the destination will be the software application that you are using to sample with. For the sake of explanation, I will assume that you are using a mixer. Without a mixer, the calibration is much simpler, so I prefer to take a harder example and work off that.

The steps to follow are quite simple and make total sense.

1. Connect the source to your mixer and attain unity gain. Unity gain is a subject that is, yet again, hotly debated by tech-heads. Basically, it means to align your sound to a fader and meter readout of 0. That is very simplistic and probably means nothing to you, so I will explain in more practical terms. Let us assume that you are connecting a synthesizer to channel 1 on your mixer. You first turn the volume knob on the synthesizer to 75%, some say crank it all the way to 100%, but I prefer to leave a little room in the event that I might need to boost the signal.

Now, you set your mixer’s fader on channel 1 to 0 and the trim post or gain pots to 0. All you now need to concentrate on is the trim/gain knob. Turn this clockwise until the meter peaks at 0dB. If you do not have VU meters on your mixer, then check the LED for that channel and make sure it does not peak beyond 0dB. If you do not have an LED for individual channels, then use the master LED for the main outs, BUT make sure that every channel but channel 1 is muted. The reason for this is that ‘live’ channels will generate a certain amount of gain or noise, even if there is no signal present, and that when you sum all the channels together, then you might get a tiny amount of gain or noise at the resultant master outs. Actually, as a general rule, when you are not using a channel, mute it, this makes for a quieter mixer.

Purists will say that peaking just past 0dB is better, but that is not the case. The reason is that mixers will sum the channels to a stereo master and even if all your faders were at 0dB, the master fader could exceed the 0dB peak. For analog mixers, that is not a problem as there is ample headroom to play with. For digital mixers, that equates to clipping.

You have now achieved unity gain. Your fader is set to 0dB and your channel’s gain/trim knob controls the gain. On some mixers, you will actually see the letter U on gain/trim knobs, helping you to identify the unity location. In essence, the knob should be at U, but that is not always the case. The second method of attaining unity gain is to do the following: Mackie mixers have a U on their trim knobs, so if you set this knob to U and your fader to 0dB, then adjust the synthesizer volume till the meter peaks at 0dB, then you have attained unity gain. I have a Mackie mixer and I always end up a couple of dBs past the U setting on the trim knobs. Don’t let this worry you. What you must try to achieve is unity gain.

Ok, so we have now set unity gain for the source and the channel input on the mixer, cool. Now we need to calibrate the mixer to the sound card.

2. Now check your master outs on your mixer. I am not talking about the control room outs that are used for your monitors but the master out faders. These will be a stereo pair. A point to make here, before we carry on, is that most people will use subgroups as the outs to the sound card’s inputs. What I have done so far is to avoid the issue of subgroups or ADAT connections because I want you to understand the straight forward signal path, and that most users have a simple mixer with limited if any, subgroups.

However, treat the explanation for the master outs as if it were for the subgroup outs. At the end of the day, they are just outputs, but the beauty of subgroups is that they can be outputted to different devices and even more important, they can have different processors like gates or compressors on each subgroup, and by assigning a channel to a subgroup, you are able to have variety in your signal path. I have 8 subgroups on my mixer and I have a different compressor inserted on each one, but I have all 8 subgroups going out and into the 8 ins on my soundcard. I can then assign a number of channels to any subgroup and use any of the compressors on them, or just have 8 outs nice and clean. The other advantage of having subgroups is that you have additional EQs that you can use. Remember that the example I am giving here, of my setup, is purely for sampling purposes as I am not sampling 8 outs at the same time.

I am sampling either a mono channel or a stereo channel and the subgroups afford me further editing and processing options. For recording purposes, I would assign my subgroups differently, but we will come to that in my new tutorial about mixing and production. For now, we are only concerned with sampling.

Back on topic: Make sure your master outs are set to 0dB.

We now have unity gain from source, all the way to the destination. What you should now be getting on your meters is 0dB at channel 1 and 0dB on the master outs.

3. The sound card settings are the one area that most people have problems with. They set their sound card faders, or gain/trim knobs, at 0dB and wonder why their levels are either coming in too low or too high. If you read part 1 of this tutorial, you will understand a little more about the processes that take place within a digital domain and the A/D input stage. All you need to concern yourself with is to have unity gain right through the signal path. So, quite simply, adjust the sound card’s faders until your meters read 0dB. Open up the software application that will be doing the recording, pass a signal through the source to the destination (the application) and check the meters within the software application. There should be no, or very little, difference in the readout.

I cannot tell you how many home studios, and even pro studios, I have been to where the signal path is not calibrated and levels are all over the place. Not attaining a calibrated path results in bad mixes, confused recordings and total frustration at not being able to understand why or what is wrong with your setup.

It is also important to mention that the minute you introduce any device into this path, you will need to calibrate to compensate for the new intruder. Compressors are the real culprits here.

I will end this month’s tutorial off with a little information on the subject of noise.

Almost all devices will produce noise, all at varying degrees. Whether it is hiss, hum or just general unwanted noise, you are left with a situation whereby you want that clean signal, noise-free. The more devices you introduce into the path, the more noise is generated. Even mixers have an element of noise, generated from their circuitry. The tried and tested trick is to use noise gates or noise filters to cut out the unwanted frequencies. Some high-end mixers will have gates built into the channels for this very purpose.

You can insert a noise gate on the master outs and adjust the parameters until you eliminate the unwanted frequencies. A gate is exactly that, a gate that opens at a specified level (threshold) and shuts (release) when set to shut. You need to set the threshold to just above the noise and set the gate to stay open for infinity or a decay time that suits you. The gate will only let signals above the threshold pass through. You have parameters such as hold, release, ratio, and attack. I do not want to go into this subject in detail as I will be covering it more fully in my other tutorial, Production and Mixing. This is purely a tip to help you to maintain a clean and strong signal path.

Relevant content:

Preparing to Sample – Using hardware samplers!

Normalisation – What it is and how to use it

Topping and Tailing Ripped Beats – Truncating and Normalising

RIAA Amps and Standards

Sampling Tools and Procedures