Often I get asked the same question about which to get; active monitors or passive monitors with a separate amplifier?

To answer this, I need to first explain the differences between the two.

It is commonly understood that active monitors simply have a built-in amp and therefore need no external amp to drive them, and that passive monitors need an external amp to drive them. Whereas this is true as far as the power is considered, it is a little more detailed than that when it comes to how each unit functions.

What we really need to look at is the crossover, which splits the signal into the appropriate frequency ranges before they’re sent to the individual drivers.

In passive designs, the monitor contains a set of passive components to split the input signal up into the various frequency bands required for each driver. The high-level input signal required to drive the speaker comes from an external power amplifier.

In active designs, the cabinet houses multiple power amplifiers connected to each driver, each amp drives a driver. The frequency band splitting is performed on the line input signal directly prior to the amplifiers.

While we are on the subject, let’s not forget the ‘powered’ monitor. Normally, in active systems, there is an amp for each driver, in powered systems, there is usually only one amp powering both drivers via a normal passive crossover.

Each design has its advantages and disadvantages.

In the case of the passive design, you are afforded a great deal of flexibility as you can choose different amps to power them and this can sometimes be a great situation to be in as the better the amp, the better the output signal. A better amp will also deliver far more headroom than a weaker counterpart and the frequency representation can also be better, especially in the higher frequency spectrum. This ‘mixing and matching’ gives the user a lot of room to try various amps and to optimise the best monitor and amp combination. It is also cheaper to buy a passive system as build costs are much lower than an active system.

In the case of the active system, the crossover can be more detailed and accurate, thus providing a more precise ‘frequency splitting’. This design also incorporates better amp matching for the drivers and therefore affords a more stable and better protected system. However, a good active system can cost considerably more than the passive counterpart.

At the end of the day, it comes down to budget, studio requirements, and space.

A passive system and separate amp take up more space than their active counterpart, but the mixing and matching of amps to monitors are very appealing and much easier to integrate into an updating studio environment. By just changing the amp, you can change the ‘colour’ and performance of the passive monitors.

Active systems come into their own when the budget starts to creep up. A good active system can actually end up being cheaper than the passive + amp alternative and can deliver better results, or rather, more precise results.

In today’s markets, the mid to upper price ranges, active systems do offer some distinct advantages. We have talked about precision and detail of amps to drive the drivers, better crossovers etc, but we also need to think about driver protection circuitry. This is as important as the drivers and amps. You tend to find that this circuit protection tends to go hand in hand with active designs. Shorter cable lengths within the cabinet, connecting amp to the driver, also negates a lot of problems that prevail due to badly shielded cables and the long lengths used.

At the budget end, things are not so rosy. Due to market competition, monitor manufacturers try to keep costs down as low as possible, and invariably compromises have to be made, and it’s usually the drivers and amps that give way.

The powered monitor will usually cost less than the active counterpart as it uses the one solitary amp to drive the drivers. So, it’s worth looking at these options before parting with your hard-earned paper.

If you consider the dynamic range of varying bit depths, 1 bit being roughly equivalent to 6dB of dynamic range, then it makes sense that the higher the bit depth the higher the dynamic range. With 24 bit depth, the dynamic range (theoretically) is 144dB. Bearing in mind that our hearing does not even come close to a 144dB range, it makes sense to use a dynamic range beyond our hearing’s dynamic range for the very simple reason that audio captured at this bit resolution will fall below our hearing’s minimum range and above its maximum range.

To accommodate internal processing within a digital system a much higher headroom is required for the simple reason that processing will require additional bits. By adding more than one 24 bit numbers together it is obvious that more bits are required. Dynamic processing, by its very nature, requires higher bit counts as the process itself generates bits, or subs of, that need managing otherwise there will be sonic compromises.

The 32 bit system seems to handle these processes well and it has become a minimum standard. Of course, we now have higher bit internal processing.

Fixed Point systems use the 32 bits in the standard way and the maths is simply a scale that provides a dynamic range of  192dB (32×6). The usual procedure is to allow the 24 bit signal to work closely at the top of the 32 bit processing. This makes complete sense as it provides a higher headroom and a lower noise floor.

Floating-point still uses the 32 bit system but arranges the bits in a different manner. The signal is still kept in 24 bit but the remaining bits are allocated to denote scaling factors. This basically means that the 24-bit can be used in a more flexible and dynamic manner allowing for a massive dynamic range. This equates to a never-ending scale of headroom and a noise floor that is so low as to be negligible.

Relevant content:

Digital Audio – Understanding and Processing

Jitter in Digital Systems

Dither – What is it and how does it work?

This is the level difference between the signal level and noise floor. The best way to describe this is by using an example that always works for me. Imagine you are singing with just a drummer. You are the signal and the drummer is the noise (ha.ha). The louder you sing or the quieter the drummer plays the greater the signal to noise ratio. This is actually very important in all areas of sound technology and music. It is also very relevant when we talk about bit depth and dynamic range.

Imagine using 24 bits. That would allow a dynamic range of 144dB (generally, 6 dB is allocated for each Bit).
Bearing in mind we have a limit of 120dB hearing range (theoretical) then the audio signal would be so much greater than the noise floor that it would be almost noiseless.

People still find it confusing to distinguish between signal-to-noise ratio and dynamic range, particularly when dealing with the digital domain.

The signal-to-noise ratio is the RMS (Root Mean Square) level of the noise with no signal applied (expressed in dB below maximum level). Dynamic range is defined as the ratio of the loudest signal to that of the quietest signal in a digital system (again expressed in decibels (dB)).

In a typical professional analog system, the noise floor will be at about -100dBu. The nominal level is +4dBu, and clipping is typically at about +24dBu. That basically equates to about 20dB of headroom and a total dynamic range of about 120+dB. Clipping in an analog system equates (when used in small stages) to harmonic distortion. This is why ‘driving’ the headroom ceiling would sometimes make the audio sound more pleasing.

Digital systems operate in finite and critical terms, and ‘driving’ the ceiling cannot be done. As digital works off a linear system, once the quantising scale is reached clipping takes place (enharmonic distortion).

Luckily, converter technology has improved so much that we now have 24 bit delta-sigma converters offering 120dB of dynamic range, similar to what we had/have in analog consoles. And by using the same methodology, by leaving ample headroom, we are able to have a great dynamic range and a strong S/N offering a negligible noise floor.

In practice, this equates to the following:
Working with a nominal level of -18dBFS (EBU) or -20dBFS (SMPTE/AES), we can attain approximately 20dB of headroom whilst keeping the noise floor about -100dB.

Digital systems cannot record audio of greater amplitude than the maximum quantising level (please read my tutorial on the Digital Process). The digital signal reference point as at the top of the digital meter scale is 0dBFS, FS standing for ‘full scale’.

In the US, the adopted standard of setting the nominal analog level is; 0dBu equals -20dBFS, thereby tolerating peaks of up to 20dBu. In Europe, 0dBu equals to -18dBFS, thereby tolerating peaks of up to +18dBu.

This all sounds complicated but all you really need to be concerned with, as far as the digital world is concerned, is that we have a peak meter scale of 0dBFS. Beyond this and you have clipping and distortion.

Relevant content:

Headroom and Dynamic Range

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

Understanding how sound travels in a given space is critical when setting up speakers in your studio.

Sound Waves  

Let us have a very brief look at how sound travels, and how we measure its effectiveness.  

Sound travels at approximately 1130 feet per second (about 1 foot per ms).
By the way, this figure is a real help when setting up microphones and working out phase values.

Now let us take a frequency travel scenario and try to explain its movement in a room.

For argument’s sake, let’s look at a bass frequency of 60 Hz.

When emitting sound, the speakers will vibrate at a rate of 60 times per second. Each cycle (Hz) means that the speaker cones will extend forward when transmitting the sound, and refract back (rarefaction) when recoiling for the next cycle.  

These vibrations create peaks on the forward drive and troughs on the refraction. Each peak and trough equates to one cycle. 

Imagine 60 of these cycles every second.

We can now calculate the wave cycles of this 60 Hz wave. We know that sound travels at approximately 1130 feet per second, so we can calculate how many wave cycles that is for the 60 Hz wave. We divide 1130 by 60, and the result is around 19 feet (18.83 if you want to be anal about it). We can now deduce that each wave cycle is 19 feet apart. To calculate each half-cycle, i.e. the distance between the peak and trough, drive and rarefaction, we simply divide by two. We now have a figure of 91/2 feet. What that tells us is that if you sat anywhere up to 91/2 feet from your speakers, the sound would fly past you completely flat.
However, this is assuming you have no boundaries of any sort in the room, i.e. no walls or ceiling. As we know that to be utter rubbish, we then need to factor in the boundaries.

These boundaries will reflect back the sound from the speakers and get mixed with the original source sound. This is not all that happens. The reflected sounds can come from different angles and because of their ‘bouncing’ nature; they could come at a different time to other waves. And because the reflected sound gets mixed with the source sound, the actual volume of the mixed wave is louder.

In certain parts of the room, the reflected sound will amplify because a peak might meet another peak (constructive interference), and in other parts of the room where a peak meets a trough (rarefaction), frequencies are canceled out (destructive interference).

Calculating what happens where is a nightmare.
This is why it is crucial for our ears to hear the sound from the speakers arrive before the reflective sounds. For argument’s sake, I will call this sound ‘primary’ or ‘leading’, and the reflective sound ‘secondary’ or ‘following’.

Our brains have the uncanny ability, due to an effect called the Haas effect, of both prioritizing and localizing the primary sound, but only if the secondary sounds are low in amplitude. So, by eliminating as many of the secondary (reflective) sounds as possible, we leave the brain with the primary sound to deal with. This will allow for a more accurate location of the sound, and a better representation of the frequency content.

But is this what we really want?

I ask this because the secondary sound is also important in a ‘real’ space and goes to form the tonality of the sound being heard. Words like rich, tight, full etc. all come from secondary sounds (reflected). So, we don’t want to completely remove them, as this would then give us a clinically dead space. We want to keep certain secondary sounds and only diminish the ones that really interfere with the sound.

Our brains also have the ability to filter or ignore unwanted frequencies. In the event that the brain is bombarded with too many reflections, it will have a problem localizing the sounds, so it decides to ignore, or suppress, them.

The best example of this is when there is a lot of noise about you, like in a room or a bar, and you are trying to have a conversation with someone. The brain can ignore the rest of the noise and focus on ‘hearing’ the conversation you are trying to have. I am sure you have experienced this in public places, parties, clubs, football matches etc. To carry that over to our real-world situation of a home studio, we need to understand that reflective surfaces will create major problems, and the most common of these reflective culprits are walls. However, there is a way of overcoming this, assuming the room is not excessively reflective and is the standard bedroom/living room type of space with carpet and curtains.

We overcome this with clever speaker placement and listening position, and before you go thinking that this is just an idea and not based on any scientific foundation, think again. The idea is to have the primary sound arrive at our ears before the secondary sound.   Walls are the worst culprits, but because we know that sound travels at a given speed, we can make sure that the primary sound will reach our ears before the secondary sound does. By doing this, and with the Haas effect, our brains will prioritize the primary sound and suppress (if at low amplitude) the secondary sound, which will have the desired result, albeit not perfectly.

A room affects the sound of a speaker by the reflections it causes. Some frequencies will be reinforced, others suppressed, thus altering the character of the sound. We know that solid surfaces will reflect and porous surfaces will absorb, but this is all highly reliant on the materials being used. Curtains and carpets will absorb certain frequencies, but not all, so it can sometimes be more damaging than productive. For this, we need to understand the surfaces that exist in the room. In our home studio scenario, we are assuming that a carpet and curtains, plus the odd sofa etc, are all that are in the room. We are not dealing with a steel factory floor studio.

In any listening environment, what we hear is a result of a mixture of both the primary and secondary (reflected) sounds. We know this to be true and our sound field will be a combination of both. In general, the primary sound, from the speakers, is responsible for the image, while the secondary sounds contribute to the tonality of the received sound. 

The trick is to place the speaker in a location that will take of advantage of the desirable reflections while diminishing the unwanted reflections. ‘Planning’ your room is as important as any piece of gear. Get the sound right and you will have a huge advantage. Get it wrong and you’re in the land of lost engineers.

Relevant content:

Sinusodial Creation and Simple Harmonic Motion

Frequency and Period of Sound

Total and Partial Phase cancellation

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

The first premise to understand is that simple harmonic motion through time generates sinusoidal motion.

The following diagram will display the amplitude of the harmonic motion and for this we need to use the term A in our formula. We will also be using θ.

I have used the equation A sin θ where θ completes one cycle (degrees).
The axis displays values based on a unit circle with being interpreted as amplitude.
The x axis denotes degrees (θ)

It then follows that:

When the angle θ is 0° or 180° then y = 0
sin 0° and sin 180° = y/A = 0

When the angle θ is 90° then y = 1
sin 90° = y/A = 1

When the angle θ is 270° then y = −1
sin 270° = y/A = −1

When constructing and working with sinusoids we need to plot our graph and define the axis.

I have chosen the y-axis for amplitude and the x-axis for time with phase expressed in degrees. However, I will later define the formulae that define the variables when we come to expressing the processes.

For now, a simple sine waveform, using each axis and defining them, will be enough.

I will create the y-axis as amplitude with a range that is set from -1 to +1.
y: amplitude

Now to create the x-axis and define its variables to display across the axis.

The range will be from -90 deg to 360 deg
x: time/phase/deg

The following diagram displays the axis plus the waveform and the simplest formula to create a sinusoid is y-sinx

The diagram shows one cycle of the waveform starting at 0, peaking at +1 (positive), dropping to the 0 axis and then down to -1 (negative).

The phase values are expressed in degrees and lie on the x-axis. A cycle, sometimes referred to as a period, of a sine wave is a total motion across all the phase values.

I will now copy the same sine wave and phase offset (phase shift and phase angle) so you can see the phase values and to do this we need another simple formula and that is:
y=sin(x-t) where t (time/phase value) being a constant will, for now, have a value set to 0. This allows me to shift by any number of degrees to display the phase relationships between the two sine waves.

The shift value is set at 90 which denotes a phase shift of 90 degrees. In essence, the two waveforms are now 90 degrees out of phase.

The next step is to phase shift by 180 deg and this will result in total phase cancelation. The two waveforms together, when played and summed, will produce silence as each peak cancels out each trough.

Relevant content:

Frequency and Period of Sound

Total and Partial Phase cancellation

Digital Audio – Understanding and Processing

So, what is Panning Law and does anyone really care?

Hell yeah!

In fact, I can state that many producers/engineers that I meet have little or no clue about the panning law.
As far as they are concerned, you turn the knob and the sound moves from one side of the stereo field to the other.

What is not understood is that certain things happen to the sound when it is panned from one side of the field, through the centre, and then to the other side of the field.

When you are dealing with monaural sounds you have to take into account the dynamics of the sound when it is moved and summed. This is very important when it comes to the mix stage as many people seem to complain about how the sound behaves when it is panned, to the point that most software developers have tried to accommodate for the panning law in their coding.

The problem facing most newcomers to this industry is that once a project is mixed in certain software and the project is then imported in separate mix software, the panned levels go for walkies. This is down to how the software behave and process the panning law, compensating for the characteristics of the process.

When a signal is panned centrally,  the same signal will be output (identically) on both the left and right channels. If you were to pan this signal from the extreme left channel through the centre and then onto the extreme right channel, it will sound as if the level rises as it passes through the centre. The panning law was integrated to introduce a 3dB level drop at the centre. If you were to sum the left and right channels in a mono situation, the centre gain would result in a 6dB rise, so attenuating by that amount became a must in the broadcast industry as mono compatibility is always a prime issue.

So, what the hell is the panning law?

The panning law determines the relationship between the sound’s apparent image position and the pan knob control. This refers to the way the sound behaves when it is moved across the stereo field. The usual requirement is that it moves smoothly and linearly across the field. This is, of course, pertinent to log/anti-log laws. If there was a linear gain increase in one channel and a linear gain decrease in the other channel to change the stereo position, at the centre position the sum of the two channels sounded louder than if the signal was panned full left or full right.
Why do you think we had to attenuate the gain whenever we panned a sound central?

Digital consoles and the digital domain started to change this thinking and accommodate and compensate for this behaviour.

It became necessary to attenuate the centre level by four common centre attenuation figures: 0, -3. -4.5 and -6dB. The -3dB figure is the most natural because it ensures that the total acoustic power output the studio monitors remains subjectively constant as the source is panned from one extreme of the stereo field to the other. However, it also produces a 3dB bulge in level for central sources if the stereo output is summed to mono, and that can cause a problem for peak level metering for mono signals. So, most broadcast desks employ a -6dB centre attenuation so that the derived mono signal is never louder than either channel of the stereo source. However, sounds panned centrally may end up sounding a little quieter than when they are panned to the edges.

Confusing huh?

Well, the answer is to simply compromise and split the difference and this is what led to most modern analogue consoles working off a -4.5dB centre attenuation.

So, what does this mean to you and how does it help you, particularly if you are working ITB (in the box) and with different software? The answer is quite simple: find out what the software panning preferences are and adjust to taste. Most of today’s software will allow for fine-tuning the panning law preferences. Cubase, along with most of the big players, has a preference dialogue box for exactly this. Cubase defaults to -3dB (classic equal power), but has settings for all the standards and I tend to work off -4.5dB.

If you stay with the old school days of 0dB, then you are in ‘loud centre channel land’, and a little bit of gain riding will have to come into play.

Check your software project and make sure you set the right preference, depending on what the project entails in terms of the mix criteria.

If you would prefer a visual explanation then try this video tutorial:

The Pan Law within your DAW explained in detail


Does what it says on the tin. By adding one or more basic and simple waveforms together and their harmonics you create a complex waveform. However, you need to add an enormous amount of harmonics to create the simplest of sounds and this type of synthesis can be complicated to create in the form of a synthesizer but the Kawai K5000 does exactly that. You can create extremely rich textures or wild and crazy sounds on this beast. Personally, I love additive synthesis but then again I am receiving very intense therapy. The process of additive synthesis is also referred to as summing the waveforms and harmonics. This method adopts Fourier analysis. Described as the representation of a sound’s frequency components as a sum of pure sinusoidal waves. An analysis of a sound’s frequency components is taken at a steady state to give an approximation of that sounds spectrum. As most natural sounds are spectrally dynamic, one single Fourier analysis could not possibly represent a sound in sine waves. By ‘windowing’, a Fast Fourier Transform (FFT) takes several of these approximations and strings them together to better predict a sound’s spectrum over time. Although this is daunting to take in it is crucial to know and I don’t expect you to understand Fourier analysis but just thought I would bring it in now as we will come back to this at the advanced stages of these tutorials.


This process involves the generating of complex waveforms and then filtering the frequencies so that you are then left with the sound you want. You take away the frequencies. Obviously the filters are crucial in subtractive synthesis and the better the filters and the wider the choice of filters available, the better the end result will be. When I teach students I find Subtractive Synthesis to be the easiest and most logical introduction to sound design.


The output of one oscillator (modulator) is used to modulate the frequency of another oscillator (carrier). These oscillators are called operators. FM synthesizers usually have 4 or 6 operators. Algorithms are predetermined combinations of routings of modulators and carriers. To really explain this I would have to go into harmonics, sidebands, non-coincident and coincident series and the relationships between modulators and carriers. So I won’t go there. What I will say is that FM synthesis can create lovely digital types of sounds, from brittle to lush. A little bit of info for DX owners, is that the oscillators on these synthesizers were all sine waves.


This form of synthesis simulates the physical properties of natural instruments, or any sound, by using complex mathematical equations in real-time. This requires huge processing power. You are not actually creating the sound but, you are creating and controlling the process that produces that sound. Waveguides and algorithms come into this process heavily but, again, I won’t go into that as it would complicate issues and confuse you. What I do find interesting is that the Nord Lead uses PM synthesis to emulate an analog synthesizer. Weird huh?


This type of synthesis takes short attack sampled waveforms called PCM, or Pulse Code Modulation and combines them with synthesized sounds that form the body and tail of the new sound. By layering these and combining them with the synthesized portion of the sound you arrive at the new sound. The final sound is processed by using filters, envelope generators etc. This is one of the most common forms of synthesis used in the 90s and even today. Roland was the most famous for adopting this type of synthesis and the D50 was one of the most common of the synthesizers that used LA synthesis. By the way, a great synthesizer and still used today.


This form of synthesis incorporates the use of pre-recorded digitized audio waveforms of real or synthetic instruments. The waveforms are then stored in memory and played back at varying speeds for the corresponding notes played. These waveforms usually have a looped segment which allows for a sustained note to be played. Using envelopes and modulators, these waveforms can be processed and layered to form complex sounds that can often be lush and interesting. The processes are algorithmic and memory is crucial to house the waveforms. I could get into linear crossfading sequentially, quasi-periodic and sine functions etc. But I won’t. I care about your sanity.


This is the method by which tiny events of sounds (grains or clouds) are manipulated to form new complex sounds. By using varying frequencies and amplitudes of the sonic components, and by processing varying sequences and durations of these grains, a new complex sound is formed. Simply put, this form of synthesis creates some crazy and interesting sounds.


This method of synthesis incorporates the combining and processing of digital waveforms. Using PCM samples, effects and filtering this method of synthesis can create stunning sounds, from lush and evolving pads to strange stepped sequences. Korg made the famous Wavestation range of synthesizers and these were based around the Sequential Circuits Prophet VS. Working off a 2-dimensional envelope using an X and Y axis (joystick) and 4 voices, this synthesizer also had wave sequencing, playing a loopable sequence of PCM samples in a rhythmic and/or crossfaded fashion. The idea was to be able to crossfade 2 or more waveforms using the joystick. Freaky but incredible.

Synthesizers used to be categorized into 2 categories, analog and digital, but in the modern world, we have the hybrid digital-analog synthesizer. Confused? Don’t be. It actually makes good sense. Whether the processing is digital or analog, the sounds are produced in much the same way, albeit with different terminology and some different routings. A good example of the terminology being adapted is: VCO, voltage controlled oscillator becomes a DCO, digitally controlled oscillator.

Of course, with today’s influx of synthesizers, be they analog, digital or a hybrid system, we are offered many varying forms of synthesis, from granular to transwave and so on.

Ultimately, synthesis is about sound design and manipulation and we now have fully fledged systems whereby effects are incorporated into the signal matrix, dynamics at the output stage and modulation matrices that make modular synthesizers look primitive. But would I swap a modular synth for a hybrid plugin? Hell no!