Frequency and Period of Sound

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When dealing with events, as we do for cycles as an example, we are concerned with two factors: Frequency (f) and Time (T). If we look at a single event then T is defined as the start to the end of that event and that amount is measured as a Period.
When dealing with a waveform cycle, the time it takes for the cycle to return to its starting position is defined as Periodicity. Taking this a step further, Frequency is then defined as the number of events that occur over a specified time, and this is illustrated with the following equation:

We measure this Periodicy in seconds (s), cycles per second. The SI unit for one cycle per second is measured in Hertz (Hz). We tend to measure anything above 1000 Hz as kHz (kilohertz) and if dealing with cycles that are measured in shorter durations than one second we use ms (milliseconds: 1/1000th of a second). This is a huge advantage when it comes to measuring microphone distances from sources and trying to correct alignments.

Relevant content:

Total and Partial Phase cancellation

Dither – What is it and how does it work?

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Dither is used when you need to reduce the number of bits. The best example, and one that is commonly used, is when dithering down from 24 bits to 16 bits or 16 bits down to 8 etc…Most commonly, dithering from a higher bit depth to a lower one takes place when a project you are working on needs to be bounced down from 24 bits to 16 bits using dithering algorithms.

So, what is the process; in mortal language of course?

A very basic explanation is we add random noise to the waveform when we dither, to remove noise. When we truncate the bits, ie, in this case, we cut down the least significant bits, and the fact that we are always left with the stepped like waveforms in the digital process, by adding noise we create a more evenly flowing waveform instead of the stepped like waveform. It sounds crazy, but the noise we add results in the dithered waveform having a lower noise floor. This waveform, with the noise, is then filtered at the output stage. I could go into this in a much deeper context using graphs and diagrams and talking about probability density functions (PDF) and resultant square waves and bias of quantisation towards one bit over another. But if I did that you’d probably hate me. All that matters is that dither is used when lowering the bit depth and that this is an algorithmic process, ie using a predetermined set of mathematical formulas.

If we take the 24 bit project scenario and select to bounce the resultant audio without dithering then the last eight bits (also know as Least Significant Bits) of every 24-bit sample are discarded. In terms of audio integrity, you will not only lose resolution but also introduce  Quantisation Noise. Because dithering adds random noise to the lower eight bits of the 24 bit signal whilst maintaining stereo separation the quantisation noise is dramatically reduced. It then makes sense to dither from 24 bits to 16 bits rather than bounce without it.

How well the process is executed is down to how good the dithering algorithms are. But to be honest these algorithms are so good nowadays that even standard audio sequencing suites (Cubase, Logic etc) will perform dithering tasks without much problem.

My recommendation is to always work in 24 bit and dither down to 16 bit for the resultant file, as CD format is still 16 bits.

Relevant content:

Jitter in Digital Systems

Digital Audio – Understanding and Processing

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AN INTRODUCTION TO DIGITAL AUDIO

In the old days, sampling consisted of recoding the audio onto magnetic tape. The audio, (analogue), was represented by the movement of the magnetic particles on the tape. In fact, a good example is cutting vinyl. This is actually sampling because you are recording the audio onto the actual acetate or disc by forming the grooves. So, the audio is a continuous waveform.

Whether we are using a hardware sampler, like the Akais, Rolands, Yamahas, Emus etc…, or software samplers on our computers, like Kontakt, EXS24, NN-19 etc…, there is a process that takes place between you recording the analogue waveform (audio) into the sampler and the way the sampler interprets the audio and stores it.. This process is the conversion of the analogue signal (the audio you are recording) into a digital signal. For this to happen, we need what we call an analogue to digital converter (ADC) and for the sampler to play back what you have recorded and for you to hear it, the process is reversed but with a slightly different structure and process, and for that to happen we need a digital to analogue converter (DAC). That is simple and makes complete sense. Between all of that, there a few other things happening and with this diagram (fig1) you will at least see what I am talking about.

Fig1

The sampler records and stores the audio as a stream of numbers, binary, 0s and 1s, on and off. As the audio (sound wave) is moving along the ADC records ‘snapshots’ (samples) of the sound wave, much like the frames of a movie.. These snapshots (samples) are then converted into numbers. Each one of these samples (snapshots) is expressed as a number of bits. This process is called quantising and must not be confused with the quantising we have on sequencers although the process is similar. The number of times a sample is taken or measured per second is called the sampling rate. The sampling rate is measured as a frequency and is termed as kHz, k=1000 and Hz= cycles per second. These samples are measured at discrete intervals of time. The length of these intervals is governed by the Nyquist Theory. The theory states that the sampling frequency must be greater than twice the highest frequency of the input signal in order to be able to reconstruct the original perfectly from the sampled version. Another way of explaining this theory is that the maximum frequency that can be recorded with a set sample rate must be half the sample rate. A good example at this point would be the industry standard cd. 44.1 kHz means that the number of times a sample (snapshot) per second is taken equates to 44,100/second.

Ok, now let’s look at Bits. We have talked about the samples (snapshots) and the numbers. We know that these numbers are expressed as a number of bits. The number of bits in that number is crucial. This determines the dynamic range ( the difference between the lowest value of the signal to the highest value of the signal) and most importantly, the signal to noise ratio (S/N). For this, you need to understand how we measure ‘loudness’. The level or loudness of a sound is measured in decibels (dB), this is the unit of measure of the replay strength ( loudness) of an audio signal. Named after this dude Bell. The other measurement you might come across is dBu or dBv, that is the relationship between decibels and voltage. This means that decibels referenced to .775 volt. You don’t even need to think about this but you do need to know that we measure loudness (level) or volume of a sound in decibels, dB. 

Back to bits. The most important aspect of bits is its resolution. Let me explain this in simpler terms. You often come across samplers that are 8 bit, Fairlight CMI or Emulator 11, or 12 bit, Akai S950 or Emu SP1200, or 16 bit, Akai S1000 or Emulator 111 etc..You also come across sound cards that have 16 bit or 24 bit etc…Each bit refers to how accurately a sound can be recorded and presented. The more bits you have (Resolution), the better the representation of the sound. I could go into the’ electrical pressure measurement at an instant’ definition but that won’t help you at this early stage of this tutorial. So, I will give a little simple info about bit resolution.

There is a measurement that you can use, albeit not clear cut but at least it works for our purposes. For every bit, you get 6dBs of accurate representation. So, an 8 bit sampler will give you 48dB of dynamic range. Bearing in mind that we can, on average, hear up to 120dB, that figure of 48dB looks a bit poor. So, we invented 16 bit cd quality which gives us a 96dB dynamic range. Now we have 24 or even 32 bit sound card and samplers (24 bit) which gives us an even higher dynamic range. Even though we will never use that range, as our ears would implode, it is good to have a bit. Why? Well, use the Ferrari analogy. You have 160mph car there and even though you know you are not going to stretch it to that limit (I would), you do know that to get to 60mph it takes very little time and does not stress the car. The same analogy can be applied to monitors (speakers), the more dynamic range you have the better the sound representation at lower levels.

To take this resolution issue a step further: 8 bits allows for 256 different levels of loudness to a sample, 16 bit allows for 65,536. So, now you can see that 16 bits gives a much better representation. The other way of looking at it is: if I gave you 10 colours to paint a painting (copy a Picasso) and then gave you a 1000 colours to paint the same painting, which one would be better in terms of definition, colour, depth etc.? We have the same situation on computer screens and scanners and printers. The higher the resolution the clearer and better defined the images on your computer, or the better the quality of the scanned picture, or better the resolution of the print. Fig2. As you can see from the figure below. The lowest bit resolution is 1 and the highest is 4. The shape of the highest bit resolution is the closest in terms of representing the shape of the audio signal above. So the higher the bit resolution the better the representation. However, remember that because we are dealing with digital processing and not a continuous signal, there will always be steps in our signal in the digital domain.

Fig2

Now let’s look at the signal to noise ratio (S/N). 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 resolution and dynamic range. Imagine using 24 bits. That would allow a dynamic range of 144 dB. Bearing in mind we have a limit of 120 dB hearing range (theoretical) then the audio signal would be so much greater than the noise floor that it would be almost noiseless.

A good little example is when people re-sample their drums, that were at 16 bit, at 8 bit. The drums become dirty and grungy. This is why the Emu SP1200 is still so highly prized. The drum sampler beatbox that gave us fat and dirty drum sounds. Lovely.

Now, let’s go back to sample rates. I dropped in a nice little theorem by Nyquist to cheer you up. I know, I know, I was a bit cold there but it is a tad relevant.

If the sampling rate is lower or higher than the frequency we are trying to record and does not conform to the Nyquist rule, then we lose some of the cycles due to the quantisation process we mentioned earlier. Whereas this quantisation is related to the input voltage or the analogue waveform, for the sake of simplicity, it is important to bear in mind it’s relationship with bits and bit resolution. Remember that the ADC needs to quantise 256 levels for an 8 bit system. These quantisations are shown as steps, the jagged shape you get on the waveform. This creates noise or alias. The process or cock-up is called aliasing. Check Fig3.

Fig3

To be honest, that is a very scant figure but what it shows is that the analogue to digital conversion, when not following the Nyquist rule, leaves us with added noise or distortion because cycles will be omitted from conversion and the result is a waveform that doesn’t look too much like our original waveform that is being recorded.

To be even more honest, even at high sampling the signal processed will still be in steps as we discussed earlier about quantisation and the way the digital process processes analogue to digital.

So how do we get past this problem of aliasing? Easy. We use anti-aliasing filters. On Fig1, you see that there are 2 filters, one before the ADC and one after the DAC. Without going back into the Nyquist dude’s issues, just accept the fact that we get a great deal of high-frequency content in the way of harmonics or aliasing with the sample rate processing, so we run a low pass filter that only lets in the lower frequencies and gets rid of the higher frequencies (above our hearing range) that came in on the signal. The filter is also anti-aliasing so it smoothes out the signal.

What is obvious is that if we are using lower sampling rates then we will need a filter that is a steeply sloped frequency band (aggressive). So, it makes sense to use higher sampling rates to reduce the steepness of the filter. Most manufacturers put an even higher sample rate at the output stage so the filter does not need to be so aggressive (please refer to upsampling further on in this tutorial). The other process that takes place is a process is called interpolation. This is an error correction circuit that guesses the value of a missing bit by using the data that came before and after the missing bit. A bit crude. The output stage has now been improved with better DACs that are oversampling, and additionally a low order analogue filter just after the DAC at the output stage. The DAC incorporates the use of a low pass filter (anti imaging filter) at the output stage.

Now let’s have a look at an aggressive form of alias called foldover. Using Nyquist again: A sampling rate of 44.1 kHz can reproduce frequencies up to 22.05kHz (half). If lower sampling rates are used that do not conform to the Nyquist rule, then we get more extreme forms of alias. Let us put that in simple terms and let us take a lower sampling rate and for the sake of this argument, let us halve the usual 44.1 kHz. So, we have a sampling rate of 22.05 kHz. We know, using Nyquist, that your sampler or sound card cannot sample frequencies above half of that, 11.025 kHz. Without the use of the filter, that we have already discussed, the sampler or sound card would still try to record those higher frequencies (above 11.025 kHz) and the result would be terrible as the frequencies would now be re-markedly different to the frequencies you were trying to record.

So, to solve this extreme form of alias, manufacturers decided to use a brick wall filter. This is a very severe form of the low pass filter and, as the name suggests, only allows frequencies at a set point through, the rest it completely omits. However, it tries to compensate this aggressive filtering by boosting the tail-end of the frequencies, set by the manufacturer, to allow it to completely remove the higher frequencies.

However, we have now come to a new improved form of DAC called upsampling.

An upsampling digital filter is simply a poor over oversampled digital reconstruction filter having a slow roll-off rate. Nowadays, DAC manufacturers claim that these DACs improve the quality of sound and when used, instead of the brick wall filters, the claim is genuine. Basically, at the DAC stage, the output is oversampled, usually 8 times, this creates higher frequencies than we had at the AC stage, so to compensate and remove these very high frequencies, a low order analogue filter is added after the DAC and just before the output. So we could have an anti-aliasing filter at the input stage and an upsampling DAC with a low order analogue filter at the output stage. This technology is predominantly used in cd players and, of course, sound cards, and any device that incorporates DACs. I really don’t want to get into this topic too much as it really will ruin your day. At any rate, we will come back to this and the above at a later date when we examine digital audio in more detail. All I am trying to achieve in this introduction is to show you the process that takes place to convert an analogue signal into digital information, back to analogue at the output (so we can hear it: Playback) and the components and processes used.

The clock. Digital audio devices have clocks that set the timing of the signals and are a series of pulses that run at the sampling rate. Right now you don’t need to worry too much about this as we will come to this later. Clocks can have a definite impact in the digital domain but are more to do with syncing than the actual digital processes that we are talking about in terms of sampling. They will influence certain aspects of the process but are not relevant in the context of this introduction. So we will tackle the debate on clocks later as it will become more apparent how important the role of a good quality clock is in the digital domain.

Dither

Dither is used when you need to reduce the number of bits. The best example, and one that is commonly used, is when dithering down from 24 bits to 16 bits or 16 bits down to 8 etc… A very basic explanation is we add random noise to the waveform when we dither, to remove noise. We talked about quantisation earlier in this tutorial and when we truncate the bits (lowering the bit resolution), ie, in this case, we cut down the least significant bits, and the fact that we are always left with the stepped like waveforms in the digital process, by adding noise we create a more evenly flowing waveform instead of the stepped like waveform. It sounds crazy, but the noise we add results in the dithered waveform having a lower noise floor. This waveform, with the noise, is then filtered at the output stage, as outlined earlier. I could go into this in a much deeper context using graphs and diagrams and talking about probability density functions(PDF) and resultant square waves and bias of quantisation towards one bit over another. But you don’t need to know that now. What you do need to know is that dither is used when lowering the bit resolution and that this is an algorithmic process, ie using a predetermined set of mathematical formulas.

Jitter

Jitter is the timing variation in the sample rate clock of the digital process. It would be wonderful to believe that a sample rate of 44.1 kHz is an exact science, whereby the process samples at exactly 44,100 cycles per second. Unfortunately, this isn’t always the case. The speed at which this process takes place usually falters and varies and we get the ‘wobbling’ of the clock trying to keep up with the speeds of this process at these frequencies. This is called jitter. Jitter can cause all sorts of problems and it is best explained, for you, as the lower the jitter the better the audio representation. This is sometimes why we use better clocks and slave our sound cards to these clocks, to eradicate or diminish ‘jitter’ and the effects caused by it. I will not go into a deep explanation of this as, again, we will come to it later in these tutorials.

So, to conclude:

For us to sample we need to take an analogue signal (the audio being sampled), filter and convert it into digital information, process it then convert it back into analogue, then filter it and output it.

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Jitter in Digital Systems

Dither – What is it and how does it work?

Active and Passive Monitors

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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.

Fixed and Floating Points

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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?

Signal to Noise Ratio – S/N or SNR

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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

RIAA Amps and Standards

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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

Sound Waves – What is Sound and how does travel in a given space?

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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

Sampling Tools and Procedures

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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

Sampling Techniques and Best Practices

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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.

Balanced

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.

Unbalanced

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