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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

Normalisation is a digital signal processing function that’s available in a lot of digital audio editing software. It scans through the program material for the highest level (Peak value), and if that level doesn’t reach the maximum available dynamic range, the software boosts the overall signal so that the Peak hits the highest level possible. For example, suppose you record a track of music and the highest peak registers at 6dB below the maximum available headroom (in this case 0). Normalisation (to 0 ceiling) brings the entire track up by 6dB. (Incidentally, most normalisation functions allow normalising to some percentage of the maximum available level; it needn’t always be 100 %.) There are a couple of problems though:

• Because normalisation boosts the entire signal, the noise floor comes up as well.

• Excessive use of amplitude-changing audio processes such as normalisation on linear, non-floating-point digital systems can cause so-called ’round-off errors’ that, if allowed to accumulate, impart a ‘fuzzy’ quality to your sound. If you’re going to normalise, it should be the very last process — don’t normalise, then add EQ, then change the overall level, and then re-normalise, for example.

If you need to normalize then think carefully about whether you will use Peak or RMS (average level).

RMS (Root Mean Square) is an averaging process. The selected audio waveform is analysed and ALL peak values are summed and divided to create an average Peak Reference which then acts as the anchor for the normalising value. In other words, the peaks are added and then divided by the number of peaks and the signal is then processed using the new average Peak value.

Peak. The selected audio is analysed and the highest Peak Value acts as the new anchor reference. All processing works from this Peak value and when normalising this Peak value is used to raise the gain to the desired limit.

I tend to find that RMS (Root Mean Square) works best on long audio files that have varying peaks and troughs. Peak tends to work well on single-shot samples, much like drum hits. Using RMS Normalisation you will often find that the processed audio will be thicker and heavier in sound whereas Peak will retain the original envelope bar louder.

If you prefer the visual approach then give this video tutorial a try:

Normalisation – what it is and how to use it

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