Other Synthesis Methods - Part 2 Physical Modeling

In music, physical modeling refers to a sound synthesis technique which is based on models of the sound production mechanisms involved in musical instruments. The idea is to generate sound by reproducing how real musical instruments actually function and produce sound.

Physical modeling is a sound synthesis technique in which the waveform of the sound to be generated is calculated using a mathematical model that simulates a physical source of sound. Engineers created the model by simplifying the law of physics involved in sound generation. The parameters that can be found in this kind of synthesizer are of two kinds; some change the constants that describe the physical materials and dimensions of the instrument, while others are time-dependent functions that describe the player’s interaction with it. (e.g. how the instrument is played by rubbing or striking or opening and closing tone holes).

This approach may seem straightforward but is in fact very different from other techniques, such as FM, additive, subtractive and sampling, which could all be referred to as 'signal-based' methods because they attempt to reproduce the output signal from an instrument without worrying about how it was produced. A physical model is obtained from the laws of physics which describe how the world around us behaves. As in other fields of physics, a physical model is nothing else than a set of mathematical equations able to reproduce what can be measured experimentally.

In the case of a guitar for example, a physical model would reproduce how the pick moves the string away from its rest position; how the string vibrates once it is released; how the string vibration is transmitted to the soundboard through the bridge; and finally how the soundboard radiates sound which we can hear

With the development of computers, scientists began to find ways to implement these models as algorithms and program them in order to produce sound. This field of research became very active in the 80's but the situation was then very different from the one we know today. It then took literally hours of number crunching on the most powerful computers of that time to obtain just a few seconds of sound. That's far from real-time! Even listening to the sound samples was not that simple as sound cards were not very common back then.
So the key factor for physical modeling has really been the increase of the power of computers which now enables us to run in real-time sophisticated enough models that can reproduce the complexity of real musical instruments.

If one looks at the music industry, Yamaha was the first company to offer a synthesizer based on physical modeling. In the early nineties, they released the VL1 which implemented physical modeling algorithms on dedicated electronics. Tassman, released by AAS in 2000, was the first software synthesizer entirely based on physical modeling.

This technology is definitely not limited to acoustic instruments. One can apply exactly the same approach to electronic instruments such as vintage synthesizers. In these cases, the computer solves in real-time models of how electric circuits used in vintage synths, filters, tube amps and effect processors functionned and behaved. The benefits are the same as for acoustic instruments. Indeed the models can reproduce the complex behavior of these electronic components resulting in sound as lively and rich as that of the hardware units.

Bibliography:

AAS - Tech Talk - Physical modeling. (n.d.). Retrieved June 24, 2016, from
https://www.applied-acoustics.com/techtalk/physicalmodeling/

Ness - Physical Modeling Synthesis (n.d.) Retrieved June 24, 2016 from
http://www.ness-music.eu/overview/physical-modeling-synthesis

Gregory Taylor - Physical Modeling Synthesis for Max Users: A Primer Published October 10, 2012, Retrieved June 24, 2016 from
https://cycling74.com/2012/10/09/physical-modeling-synthesis-for-max-users-a-primer/#.V20bE1fmpuU

Hiller, L.; Ruiz, P. (1971). "Synthesizing Musical Sounds by Solving the Wave Equation for Vibrating Objects". Journal of the Audio Engineering Society.

Karplus, K.; Strong, A. (1983). "Digital synthesis of plucked string and drum timbres". Computer Music Journal (Computer Music Journal, Vol. 7, No. 2) 7 (2): 43–55. doi:10.2307/3680062. JSTOR 3680062.

Julius O. Smith III (December 2010). Physical Audio Signal Processing.
Cadoz, C.; Luciani A; Florens JL (1993). "CORDIS-ANIMA : a Modeling and Simulation System for Sound and Image Synthesis: The General Formalism". Computer Music Journal (Computer Music Journal, MIT Press 1993, Vol. 17, No. 1) 17/1 (1).

Other synthesis methods Part 1 - Frequency Modulation (FM)

Yamaha DX7 The most famous synthesizer of the 1980s. Its electric piano became a standard sound in ballads and "smooth jazz" genres.  Its bass was the standard bass sound, typically played in bouncy octaves.  Its crystalline timbres were such a departure from the world of analog, that this synth was a super-hit for Yamaha in 1983, and spanned a long family of FM-based products.

FM is a form of synthesizer developed in the early 1970s by Dr John Chowning of Stanford University, then later developed further by Yamaha, leading to the release of the now-legendary DX7 synthesizer: a popular source of bass sounds for numerous dance musicians.

Unlike analogue, FM synthesizers produce sound by using operators, which are very similar to oscillators in an analogue synthesiser but can only produce simple sine waves. Sounds are generated by using the output of the first operator to modulate the pitch of the second, thereby introducing harmonics Like an analogue synthesizer, each FM voice requires a minimum of two oscillators in order to create a basic sound, but because FM only produces sine waves the timbre produced from just one carrier and modulator isn't very rich in harmonics

FM synthesis is based on two key things – a ‘modulator’ oscillator, and a ‘carrier’ oscillator. These oscillators usually both use a sine waveform, and from this the modulator oscillator works just like an LFO – because it modulates the frequency/pitch of the carrier oscillator. You can try this yourself on a normal subtractive synthesizer, by setting up a sine wave oscillator and an LFO, and using the LFO to modulate the pitch of the oscillator – as you increase the rate of the LFO, the sound becomes non-harmonic. Note that in FM synthesis, the word ‘oscillator’ is often replaced with the term ‘operator’. As you change the modulation of the carrier operator, the frequency of the carrier will constantly move up and down depending on how the modulator is set up (e.g. it’s depth and rate), and in doing this different harmonics are created (called ‘sidebands’), because these harmonics surround the carrier frequency depending on how it is modulated.

Because of the somewhat lifeless sound of the operators, FM synthesizers tend to include somewhere around 4-8 operators on a synth to spice things up. These extra operators can be routed in all sorts of different and interesting ways, called ‘algorithms’. For example, with the addition of an extra modulator operator, we can arrange the operators so that they go ‘’modulator 1’ & ‘modulator 2’ go into the carrier’, or ‘modulator 1 goes into modulator 2, which goes into the carrier’ – this being more complicated and creating a new waveform. Therefore, using many operators can produce unique and lifelike sounds unachievable with other types of sound synthesis.

In telecommunications and signal processing, frequency modulation (FM) is the encoding of information in a carrier wave by varying the instantaneous frequency of the wave. This contrasts with amplitude modulation, in which the amplitude of the carrier wave varies, while the frequency remains constant.

Due to the nature of FM, many of the timbres created are quite metallic and digital in character, particularly when compared to the warmth generated by the drifting of analogue oscillators. Also due to the digital nature of FM synthesizer the facia generally contains few real time controllers. Instead, many numerous buttons adorn the front panel forcing you to navigate and adjust any parameters through a small LCD display.

Notably, although both FM and analogue synthesisers were originally used to reproduce realistic instruments, neither can fabricate truly realistic timbres. If the goal of the synthesiser system is to recreate the sound of an existing instrument, this can generally be accomplished more accurately using digital sample based techniques.

if you want to know more about synthesis techniques please read. Rick Snoman's Dance Music Manual (Second Edition) Tools, Toys and Techniques.

The science of Synthesis - Part 4 Subtractive Synthesizers

Subtractive synthesis is the basis of many forms of synthesizers and is commonly related to analogue synthesis. It is achieved by combining a number of sounds or 'oscillators' together to create a timbre that is very rich in harmonics.

Having looked into the theory of sound, we can look at how this relates to a synthesizer. Subtractive synthesis is the basis of many forms of synthesizers and is commonly related to analogue synthesis. it is achieved by combining a number of sounds or 'oscillators' together to create a timbre that is very rich in harmonics.

Layout of a basic synthesizer

This rich sound can then be sculpted using a series of 'modifiers'. The number of modifiers available on a synthesizer is entirely dependent on the model, but all synthesizers offer a way of filtering out certain harmonics and of shaping the overall volume of the timbre.

The next part of this series of posts looks at how a real analogue synthesizer operates, although any synthesizer that emulates analogue synthesis (i.e. digital signal processing (DSP) analogue) will operate in essentially the same way, with the only difference being that the original analogue synthesizer voltages do not apply to their DSP equivalents.

An analogue synthesizer can be said to consist of three components.

• An oscillator to make the initial sound.
• A filter to remove frequencies within the sound.
• An amplifier to define to overall level of the sound.

Each o these components and their role in synthesis are discussed in the sections below. 

Voltage controlled Oscillator (VCO)

When a key on a keyboard is pressed, a signal is sent to the oscillator to activate it, followed by a specific control voltage (CV) to determine the pitch. The CV that is sent is unique to the key that is pressed, allowing the oscillator to determine the pitch it should reproduce. For this approach to work correctly, the circuitry in the keyboard and the oscillator must be incredibly precise in order to prevent the tuning from drifting, so the synthesizer must be serviced regularly. In addition, changes in external temperature and fluctuations in the power supply may also cause the oscillator's tuning to drift.

This instability gives analogue synthesizers their charm and is the reason why many purists will invest small fortunes in second-hand models rather than using the latest DSP-based analogue emulations. Although, that said, if too much detuning is present, it will be immediately evident and could become a major problem! There is still an ongoing argument over whether is possible for DSP oscillators to faithfully reproduce analogue-based synthesizers, but the argument in favor of DSP synthesizers is that they offer more waveforms and do not drift too widely, therefore prove more reliable in the long run.

In most early subtractive synthesizers the oscillator generated only three types of waveforms: square, sawtooth and triangle waveforms. Today this number has increased and many synthesizers now offer additional sine, noise, tai-saw, pulse and numerous variable wave shapes as well.

Although these additional waveforms produce different sounds, they are all based around the three basic wave shapes and are often introduced into synthesizers to prevent mixing of numerous basic waveforms together, a task that would reduce the number of oscillators.

For example, a tri-saw wave is commonly a sample of three sawtooth waves blended together to produce a sound that is rich in harmonics, with the advantage that the whole sound is contained in one oscillator. Without this waveform it would take three oscillators to recreate this sound, which could be beyond the capabilities of the synthesizer. Even if the synthesizer could utilize three oscillators to produce this one sound, the number of available oscillators would be reduced. 

Here concludes the fourth part of this post, if you want to know more about acoustic science, please read Rick Snoman's Dance Music Manual (Second Edition) Toys, Tools and Techniques.

Mixing : Reverb

Reverb is hundreds and hundreds of delays. When a sound first occurs, it travels throughout the room at the snail's pace of around 740 miles per hour. It bounces of the walls, ceiling and floor and comes back to us as hundreds of different delay times. All of these delay times wash together to make the sound we know as reverb.

Reverb is actually like placing hundreds of spheres of sound between the speakers. It takes up a tremendous amount of room in this limited space between the speakers. In a digital reverb, all of these delays are panned to virtually hundreds of different places between the speakers. This is why reverb masks other sounds so much in the mix.

There are certain parameters of control found in units that create reverb. I will explain each setting

Room Types

Modern digital reverbs allow the user to change the "type of room." You can simply imagine different types of rooms between the speakers. There are no strict rules as to the type of room that is used in a mix. Some engineers prefer a plate reverb sound on the snare drum. Some use hall reverbs on saxophones.

It is best to always set the type of reverb while in the mix (with all the sounds on) to make sure it cuts through the mix like you want it to. Different types of sounds will mask the reverb in different ways.

Reverb time

You can also change reverb time: the duration or length of time it lasts.

A common rule is to set the reverb time on a snare drum so that it ends before the next kick lick; this way, the snare reverb does not obscure the attack on the next kick note, which will keep the kick drum sounding clean, punchy and tight. The faster the tempo of a piece, the shorter the reverb time. Again though, rules are made to be broken. (You won't go to jail for this one.)

Pre delay time

When a sound occurs, it takes awhile for the sound to reach the walls and come back. The time of silence before the reverb begins is called the predelay time. On many units it is just called delay.

Different-sized rooms will naturally have different pre delay times. A medium-sized auditorium has around 30ms of pre delay time, while a coliseum might have as much as 100ms of pre delay time. Therefore, it is important to have a bit of pre delay time if you are looking for a truly natural reverb sound. Most times, when you call up a preset in a reverb unit, someone has already programmed a pre delay time. You can adjust this as desired.

The cool thing about longer predelay times (over 60ms or so) is that they help to separate the reverb from the dry sound. With shorter predelay times, reverb will very quickly "mush up" the original dry sound, making it unclear. With longer predelay times, a vocal, for example, will remain clean and clear even with a good amount of reverb. When using extremely long predelay times, it is important to set the delay time to the tempo of the song (as was covered when we discussed delays).

Diffusion

In most effect units, diffusion is the destiny of echoes that makes up the reverb. Low diffusion has fewer echoes.

You can actually hear the individual echoes in a low-difussion setting. It sounds kind of like "Will, il, il, il, il, bur, bur, bur, bur, bur, bur." A hall reverb setting is preset with a very low-diffusion setting. High-diffusion has more echoes - so many that they meld together into an extremely smooth wash of reverb. Plate reverbs often have a very high-diffusion preset.

There are no strict rules of rat use of high- or low-diffusion settings. High-diffusion settings tend to be sweeter, smoother, and more silky, Low-Diffusion tends to be more intense. Some engineers prefer a low-diffusion setting on a snare drum to make it sound more raucous for rock and roll. High-diffusion is often used to make vocals sound smoother.

EQ of Reverb

You can equalize reverb at various points in the signal path. First you can EQ the reverb after the signal comes back into the board (if you are using channels for your reverb returns that have EQ on them). It is usually better to use the EQ in the reverb unit itself.

Not because it is necessarily a better EQ, but because in some units you can place the EQ before or after the reverb. Ideally it is best to EQ the signal going to the reverb. If your reverb unit does not have this capability, you can actually patch in an EQ, after the master auxiliary send, on the way to the reverb unit. The truth is I don't like to EQ reverb because it screws up the natural sounds (which is just fine) should you use EQ on a sound. Sometimes EQ might be used to simply roll off some low-frequency rumble. Normally, if your reverb sounds like it needs EQ, it is often better to go back and EQ the original sound that is going to the reverb.

High- and Low-Frequency Reverb Time

Even better than using EQ on your reverb is to set the duration of the highs and lows. Many reverb units have this setting these days. This is a bit different than EQ, which changes the volume of the frequencies. High- and low-frequency reverb time changes the time that each frequency range lasts. Using these settings will generally make the reverb sound more natural than any type of EQ.

Regardless of whether you EQ your reverb or set the duration, there is a huge difference as to how much space it takes up in the mix-and the resulting masking it creates. Remember that low-frequency sounds take up way more space than high-frequency sounds. And because reverb is also hundreds of sounds, reverb with more low frequencies will take up an enormous amount of space in a mix.

Reverb with more high frequencies still takes up a lot of space, but not nearly as much as when lows are present.

Reverb Envelope

Another setting of reverb is the "envelope"; that is, how the reverb changes its volume over time: Normal reverb has an envelope where the volume fades out smoothly over time.

Engineers (being the bored people they are) thought to put a noise gate on this natural reverb, which chops it off before the volume has a chance to fade out. Therefore, volume stays even, then stops abruptly.

You can put a noise gate on your reverb, but it's much simpler to use the gated reverb settings on your effects unit. If we were to reverse the envelop of normal reverb, the volume would rise then stop abruptly.

If you take the tape, play it backward, add normal reverb, record it on open tracks on the multitrack, and turn the tape around to run forward, you'd get an effect commonly called preverb.

This effect is the most evil one that can be created in the studio; only the devil could put an effect on something before it happens. Furthermore, it has been used in every scary movie made, including The Exorcist and Poltergeist. And of course, it is one of Ozzy Osbourne's favorite effects.

One of reverb's main functions is to connect sounds in a mix and fill in the space between the speakers. 

Like any sound, reverb can be panned in various ways.

Reverb can be spread to any width by how far left and right you pan the reverb return channels on your mixing board. Depending on how you have your effects patched back into your console, or how your computer plug-ins work, you may not have this option.

Final notes:

Reverb can also be brought out front with volume...

....placed in the background by turning down the volume...

....or raised or lowered a bit with EQ.

You can read more about the most important things you need to know about mixing, and how to setup the compressors properly depending on the task you want to realize in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.

Mixing : Delays - Part 2

You need to learn how each delay time feels and what feelings or emotions each delay time evokes. Then, when you hear a song that has a similar feeling or emotion you will know which delay time might work.

Different delay Times

Let's define specific delay time ranges so that you can get to know them and incorporate them into your memory time banks.

More than 100ms

Professional engineers refer to this length of delay as echo. However the real world (and my mom) use the term echo to refer to reverb. For our purposes, we will use echo to refer to a delay time greater than 100ms, not reverb.

When setting a delay time greater than 100ms, it is important that the delay time fits the tempo of the song, otherwise it will throw off the timing of the song. The delay time should be in time, a multiple of, or an exact fraction of the tempo. If you know the beats per minute of the song, the following chart gives the relationship between tempos and delay times.

                               Beats Per Minute                   Time Between Beats

                               60   bpm                                1000    ms

                               90   bpm                                 666.6  ms

                              120  bpm                                 500     ms

                              150  bpm                                 400     ms

                              180  bpm                                 333.3  ms

                              210  bpm                                 285.7  ms

                              240  bpm                                 250     ms

Tip: / If you know the tempo of the song, you can figure out the delay time with the following formula:

Delay time = 60,000 / beats per minute.

Then any fraction or multiple of that delay time will also fit the tempo of the song. For example if the tempo is 100 BPM then the 600 ms would fit the tempo. But 150 ms, 300 ms, and 1200ms would also fit.

If you don't know the beats per minute (bpm) of the song, use the snare drum (or some other instrument playing on a continuous patterns) to set the delay time. Even if you are going to put the delay on the vocals, for example, put the delay on the snare to set the delay time to the tempo.Again,once you have found a delay time that works, any multiple or fraction of that time might also work.

A delay time over 100ms creates a dreamy effect and is most commonly placed in songs with slower tempos where there is room for the additional sound. Therefore, the more instruments and the more notes in a mix, the less often this delay time is used. This is pretty obvious - if you have no room in the mix don't add more sounds. This is especially true when there is feedback on a long delay time. The delays take up so much space in a mix that they are often only turned up at the ends of a line, where there is enough space to hear echoes by themselves.

Tip:  / Feedback is created by feeding back the delayed signal into the input, so the sound repeats, repeats, repeats.

60 to 100 ms

You can hear this delay time, commonly referred to as slap, on the vocals of Elvis Presley and in rockabilly music. In fact, there is about an 80ms delay between the syllables "rock" and "a" in the word "rockabilly."

This effect can be quite helpful in making a thin or irritating sound (especially for a voice) sound fuller. It can help obscure bad vocal technique or pitch problems. In fact, a slap can be used to bury any bad sound. However, you never want to bury anything to deep. Add too much delay on a bad vocal and not only do you have a bad vocal, but you also have a bd mix. On the other hand a sea can make a vocal seem less personal. If you have an incredible singer, you might forego using any delays. Just put it out there with a little reverb and let it shine.

30 to 60ms

Put your lips together and blow a raspberry, this sound technically called a "motorboat." The time between each flap on your lips is approximately 50ms. Delay time in this range is referred to as "doubling" because it makes a sound seem like it was played twice, or doubletracked. When a part is sung or played twice, there will naturally be a time delay ranging from 30 to 60ms (no one can ever sing or play a part twice exactly in this time). Therefore adding a delay of this length makes it sound like the part has been played twice. The beatles used this effect extensively to simulate more vocals and more instruments.

Just like a slap, doubling helps to obscure a bad sound or bad performance. So it can be used to help bury things in the mix.

Likewise, since it does obscure the purity and clarity of a sound, you should use it selectively, depending on the sound, song and style of music.

Tip: / Although doubling makes a sound seem like it has been played twice, it is a different sound than if you actually doubletrack a sound. In fact doubling often sounds so precise that it sounds somewhat electronic. This is especially true on vocals and simple sounds. However, if a sound is complex, especially if the sound is a combination of sounds (like a bunch of background vocals or a guitar sound with multiple pics), then you don't notice the precision of the delay. Therefore when you put doubling on 20 vocals, it sounds like 50 vocals and it sounds incredibly natural.

1 to 30ms

An unusual thing happens with this type of delay, commonly known as fattening. At the delay time, our brain and ears are not quick enough to hear two sounds; we only hear one fatter sound.

The threshold between hearing one sound or two sounds actually varies depending on the duration of the sound being delayed. Also the farther the sounds are panned separately, left and right the shorter the delay time before you hear two sounds. For example a guitar panned to the center with a delay in the center might require at least 40 milliseconds to hear two sounds; whereas, if the guitar and delay are panned left and right, you might hear two sounds beginning around 20 milliseconds. The following chart gives approximate thresholds for some instruments with different durations (actual thresholds will depend on the particular timbre and playing style of the instrument):

                        Approximate thresholds between hearing one sound versus two

                                                      Hi - hat                  10ms

                                                      Percussion             10ms

                                                      Snare                      15ms

                                                      Kick Drums           15ms

                                                      Piano                      20ms

                                                      Horns                     20ms

                                                      Vocals                    30ms

                                                      Guitars                   30ms

                                                      Bass Guitars          40ms

                                                      Tubas                     40ms 

Besides reverb, fattening is the most-used effect in the studio, mostly because it doesn't sound much like an effect. Fattening is the primary effect used to make a sound stereo, which has a certain magic to it. When you put the original "dry" instrument sound in one speaker and put a delay less than 30ms in the other speaker, it "stretches" the sound in stereo between the speakers.

Fattening can make an already beautiful acoustic guitar or piano sound incredible. Fattening is very effective in making a thin or irritating sound fatter and fuller. It also appears to make a sound more present simply because when a sound is in stereo, it takes up more space between the speakers. This is especially effective when you want to turn a sound down in the mix but still have it be discernible.

You have to be careful with fattening, though, because it uses up your space between the speakers. Fattening will make a mix fuller and denser, so you must make sure there is enough room between the speakers. Therefore, fattening is used most often when there are fewer notes and sounds in the mix. On the other hand if you want to create a wall of sound, even if the mix is already busy you can add fattening to make it more busy. (This blow's people's minds.) This is commonly done in heavy metal, alternative rock and some new age music.

0 to 1ms

This soft of a delay time causes phase cancellation. I will address only the critical aspects of phase cancellation here. But keep in mind that phase cancellation is a very serious problem in recording and I highly recommend that you do further research to gain a complete and clear explanation of the problem it causes.

Phase cancellation happens when two of the exact same sound, like those created with two mins or two speakers, are a little bit out of time. One example is when you switch the positive and negative wires on one of the two speakers. Now, one speaker is pushing out while the other is pulling in. When a speaker pushes out, it creates denser air than normal. When a speakers pulls in, it creates more spaced out air than normal (rarefied air). When the denser air from one speaker meets the spaced-out air from the other speaker, you end up with normal air, normal air equals silence. This means you could have two speakers blasting away and theoretically you could hear nothing.

There are many companies now using phase cancellation to quite the world. This technology is used in automobiles, on free-ways (instead of cement walls on the sides of the freeways), in factories, and even in headphones to cancel out sounds around you. Marriage counselors are selling them by the dozens.

If you have two mins on one sound at two different distances, one mic might be picking up denser air while the other mic is picking up spaced-out air. Put the two mins together in the mix and they will tend to cancel each other out, though not completely. 

Phase cancellation degrades the sound quality in the following ways.

• Loss of volume. You lose volume when both miss are on, especially when you're in mono (which, by the way, is one of the best ways to detect phase cancellation - put the board in mono or pan both sounds to the center).
• Loss of Bass. You lose bass frequencies, making the sounds thin.
• Loss of image. Most importantly, you lose the clarity and precision of the perceived image of the sound between the speakers. The sound seems to be more "spacey." Though some people like this effect, most people are addicted to clarity these days. If the mix is even played back in mono (as on TV or AM radio), the sound will disappear completely.

There are many ways to curb phase cancellation. The primary way is to simply move one of the pics. If both miss are picking up the sound in the same excursion of the wave, there will be no phase cancellation.

2 Mics picking the sound in phase

It takes 1ms for a complete wave of 1000Hz to pass by us. If you were to set a delay time of 0.5 on a sound, it would put it out of phase. Therefore, you can use a digital delay to put the sound back in time.

Finally you can remove a large amount of phase cancellation through isolation. Often, the bleed of a sound into a second mic will cause phase cancellation with the first mic. By using baffles or noise gates, you can reduce the bleed in the second mic, voiding the phase cancellation. 

You can read more about the most important things you need to know about mixing, and how to setup the compressors properly depending on the task you want to realize in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.

Mixing : Delays part 1

Time-based effects include all processes where some form of manipulation of time occurs to the signal. This includes things like delays and echos (obvious time manipulation), chorusing and flanging (short delays with modulation), phasing (shifting signals by very small amounts of time), reverbs (essentially numerous delays and echoes), pitch transposers and Harmonizers (slowing or speeding the signal while adding or removing slices to keep it in time with the music), etc. These effects all change the signal’s timing in one way or the other to produce the desired result, thus they are time-based effects.

Delays

After many failed attempts to use outdoor racquetball courts to create delays, engineers realized they could get a delay from a tape player. You could hear a delay by recording a signal on the record head, then listening to the playback head two inches later. The delay time could be set by changing the tape speed. Engineers used this technique for years. There was also a popular unite called the Echoplex, which fed a piece of tape through a maze of tape heads at different distances, each giving different delay times. Not bad, but the problem with tape is that every time you record over it, you get more tape hiss.

Then came analog delays, which would put a signal through a piece of electronics to delay the signal a bit. The more you put the signal through the electronics, the longer the delay. It was a bucket brigade type of system. The only problem was that when you put a signal through a piece of electronics over and over, it also got extremely noisy after awhile.

Then came digital delays, which record the signal digitally onto a chip, then use a clock to tell the unit when to play the sound back. 

Delay times versus distance 

Before we explore different delay settings, it is helpful to understand the relationship between delay time and distance. Sound travels at approximately 1130 feet per second. That's around 740 miles per hour, which is extremely slow compared to the speed of sound in wires - 186,000 miles per second, the speed of light (approximately 670 million miles per hour). Therefore, it is easy to hear a delay between the time a sound occurs and the time it takes for a sound to travel even a few feet to a wall and back. We can also easily hear a delay when we put two microphones at two different distances from one sound. In fact, chasing the distance between two microphones is almost exactly like chasing the delay time on a digital delay.

The following chart illustrates how different distances relate to delay time. Of course, if you are calculating a delay time based on the distance between a sound source and a wall, the distance must be doubled (to and from the wall). 

                                                               Feet     =      Delay(ms)

                                                               1130            1000

                                                               560                500

                                                               280                250

                                                               140                125

                                                               70                 62.5

                                                               35               32.25

                                                               17.5            16.13

                                                               8.75              8.01  

                                                               4.28                   4

                                                               2.14                   2

                                                               1.07                   1

As distances become smaller and smaller, the distance in feet almost equals the milliseconds of delay. This correlation comes into play when using more than one mic on a sound (e.g., piano, guitar amps, acoustic guitars, horns, or background vocals) and is especially helpful when miking drums. For example, the distance you place overhead miss above the drum set will create a corresponding delay time between the overhead miss and the snare mic (or any of the rest of the mics for that matter). It is also important to note the distance between instruments when miking an entire band live (or recording everyone in the same room at once) since miss may be more than 10 feet away from another instrument and still pick it up.

Besides delay time, you must also consider phase cancellation, a problem that happens with extremely short delay times. We'll discuss more about it later.

if you pay attention to the way that something sounds when miked at different distances, you will eventually become aware of what different delay times sound like. Once you become familiar with the way that different delays affect different sounds, you can control their use in a way you deem most appropriate; that is you can do whatever you want.

You need to learn how each delay time feels and what feelings or emotions each delay time evokes. Then, when you hear a song that has a similar feeling or emotion, you will know which delay time might work.

You can read more about the most important things you need to know about mixing, and how to setup the compressors properly depending on the task you want to realize in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.

Mixing : Panpots and Stereo Placement

When mixing, you use panpots (balance knobs) to place each sound and effect left to right between the speakers. A panpot is actually two volume controls in one. When you pan to the left the signal going to the right is turned down. When you pan to the right, the volume of the signal going to the left is turned down.

Panning in a mix is mapped out visually as a function of left to right. Panning a sound to one side or the other also seems to make the instrument a bit more distant in the mix. If the sound if panned to the center, it will seem to be a bit closer, a little more about front.

If you think of the space between the speakers as a pallet on which to place instruments left to right, the main objective might be to place each sound in a different place so you can hear each sound more clearly. However, certain styles of music have developed their own traditions for the particular placement of each instrument left to right im the stereo field. Normally, the placement of a sound is static; it stays in the same places throughout the mix. However, the movement of a pant during a mix creates an specially dramatic dynamic.

You can read more about the most important things you need to know about mixing, and how to setup the compressors properly depending on the task you want to realize in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.

Mixing : Equalizers

Eq it is one of the least understood aspects of recording and mixing probably

EQ is a change in the volume of a particular frequency of a sound, similar to the bass and treble tone controls on a stereo. it is one of the least understood aspects of recording and mixing probably because there is such a large number of frequencies - from 20 to 20,000 HZ. The real difficulty comes from the fact that boosting or cutting the volume of any one of these frequencies depends on the structure of the sound itself: Each one is different. But even more complex is the fact that different sounds are equalized differently depending o the type of music, the song, end even the people you are working with.

First you m st learn all the frequencies or pitches by name. Then, you will see how boosting or cutting a certain frequency affects different instruments in different ways.

Types of equalizers

There are three main types of equalizers found in the recording studio: graphics, parametrics, and rolloffs (highpass and lowpass filters).

Graphics

Each frequency can be turned up or down by using the volume sliders on a graphic equalizer. The voume controls on an equalizer are called bands. There are different kinds of graphic equalizers that can divide frequencies from five bands up to 31 bands. Five to 10 band graphic equalizers are commonly found in car stereos. Thirty-one band graphics (which will change the volume at 31 different frequencies) are common in recording studios and live sound reinforcement.

Band Graphic EQ

The primary advantage of a graphic equalizer is that you can make changes in a volume at a number of different frequencies. Graphic EQs got their name from the visual display that's easy to read for reference. (However , these days, you get a much nicer display on a digital parametric EQ.) Also, since the frequencies are mapped out visually from left to right, it is easy to find and manipulate the volume of any particular frequency.

Many people don't realize that when you turn up a particular frequency from a graphic, you are actually turning up a range of frequencies preset by the manufacturer. For example if you turn up 1000 hz, you might actually be turning up a frequency range from around 300 to 50000Hz.

Bandwidth on a graphic EQ

This range of frequencies is called the bandwidth and is preset by the manufacturer. You have no control over the bandwidth on a graphic. Generally, the more bands (or volume controls) there are, the thinner the bandwidth. Therefore a 31-band graphic EQ will have a more precise frequency range for each slider than a 5-band graphic. If you turn the volume of 1000Hz on a 5-band graphic you could be turning up from 100 to 10,000Hz.

Frequency works as a function of up and down, so higher frequencies are reproduced by the tweeters in our speakers and these are placed higher than the subwoofers who usually go at the bottom.

Parametrics

Engineers want to be able to control the range of frequencies, or bandwidth, they are turning up or down. With a parametric, the bandwidth knob gives you control over the width of the frequency range being manipulated. The knob is usually called "Q" because the word "bandwidth" won't fit on the knob ("Q" stands for "quality", which is an electrical term for bandwidth). A thin bandwidth is normally labeled with a peak, whereas a wide bandwidth is often labeled with a hump. Sometimes ranges of musical octaves are used to define the bandwidth - for example from 0.3 octaves to 3 octaves wide. Sometimes a scale of 1-10 or 10-1 (it's not standardized on consoles) is used.

Wide and narrow Bandwidths on Parametric EQ

On a graphic equalizer, you select the frequency by moving your arm left to right to place your hand on the correct volume slider. On a parametric EQ, you select the frequency by turning the "frequency sweep" knob with two fingers. A separate volume knob is then used to turn the chosen frequency up or down.

Paragraphics

Some less expensive consoles have equalizers with frequency sweep knobs but do not have bandwidth knobs. This type of equalizer is commonly referred to as semi-parametric, or paragraphic. Be careful, though; these days some manufacturers and certain salespeople are now using the term "parametric" to refer to a paragraphic or semi-parametric, even though it has no bandwidth control.

Rolloffs

A rolloff EQ turns down the volume of low or high frequencies. They are commonly found on consoles as highpass and lowpass filters. Larger consoles often have sweepable or variable rolloff knobs so that more of the lows or highs are rolled off. Smaller consoles often have only a button that rolls off a preset amount of lows or highs. A highpass filter rolls off the low frequencies but does nothing to the highs; it passes them.

Highpass filters are especially helpful in getting rid of low-frequency sounds, such as trains, planes, trucks, air conditioners, earthquakes, bleed from bass guitars or kick drums, and serious foot stomping.

Highpass filters can be found on microphones and smaller mixing consoles as switches that simply roll off the lows when the switch is engaged.

A lowpass filter rolls of the hight frequencies and is especially helpful in getting rid of hiss on sounds that don't have a lot of highs in them, such as bass guitar. Lowpass filters are also used to roll off the high frequency attack (click) of a kick drum in order to make it sound more like the classing rap kick drum sound.

You can read more about the most important things you need to know about mixing, and how to setup the compressors properly depending on the task you want to realize in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.

  

Mixing : Volume controls part 3 - Noise gates

Operating similarly to a compressor / limiter, noise gates are often packaged together in one box with them, know the differences

Like compressors / Limiters, a noise gate turns the volume down. The difference is that a compressor / limiter turns the volume down above the threshold, while a noise gate drops the volume when the volume falls bellow the threshold. However, since the volume is being turned down on a sound that is already low in volume, normally a noise gate will turn off the sound completely.

Noise gates have three main functions: to get rid of noise, to get rid of bleed, and to shorten the duration of a sound.

Noise Eradication

The first function of a noise gate is to get rid of noise, hiss or anything annoying that is low in volume. Noise gates only get rid of background noises when a sound is not playing. Noise gates don't get rid of noises when a sound is not playing. Noise gates don't get rid of noises while the main signal is present; however you normally can't hear the noise when the sound is playing.

One function of a noise gate is to get rid of amp noise when a guitar is not playing. Say you have a guitar amp set on 11 with lots of distortion. When not playing, the amp makes a huge "cushhhhhh" sound (when the guitar is playing, you don't hear the amp noise, it gets cut off. Whenever the guitar player is not playing, you now hear silence.

It is important not to chop off any of the guitar sound. All it takes is for the musician to play a soft note, and the noise gate will chop the sound right off. Noise gates can also be used to get rid of noise from tape hiss, cheap effects units, dogs, kids, and crickets. 

Bleed eradication

Another common use of a noise gate is to remove the bleed from other instruments in the room. When a mic is on an instrument, the sound of that instrument will be loudest in the microphone. Therefore, it is easy to set the threshold of a noise gate between the sound and the bleed, so that the bleed gets turned off.

The obvious advantage of isolating a sound like this is that you have more individual control over volume, equalization, panning and effects. Once a sound is isolated with a noise gate, any changes you make with a sound manipulator will only change the one sound you are working on. Gates can be especially effective on drums to isolate each drum. This is especially important on a snare if you are going to put a lot of reverb on it. Without the gate you end up with reverb on the hit as well. Another advantage of isolation is that it helps to eliminate phase cancellation (we'll discuss this more later).

But most importantly, by removing the bleed, you will then hear the sound in only one microphone. This has the effect of putting the instrument in one precise spot between the speakers, instead of being spread in stereo. For example, consider the miking of a hi-hat cymbal. Besides being picked up by the hi-hat mic, the hi-hat is also being picked up by the snare drum mic. If the hi-hat mic is panned to one side and the snare mic (with the hi-hat bleed) is panned to the center, the hi-hat appears to be spread in stereo between the speakers. It is no longer clear and distinct at a single spot in the mix. Putting a noise gate on the snare mix turns off the hi-hat when the snare is not playing. The isolated image of the hi-hat when the snare is not playing. The isolated image of the hi-hat in it's own mic, you will now appear to be the crystal clear and precisely defined wherever the hi-hat mic is placed in the mix.

Shortening the Duration

You can also use the noise gate to shorten the duration of a sound. The noise gate will cut off both the attack and release of a sound because these are commonly the softest parts of the sound. This can be quite an unusual effect.

A noise gate can also be put on reverb to chop off the release resulting in the well-known effect referred to as gated reverb.

Visually, when volume is shown as front to back and the volume is lss than the threshold setting, the sound will disappear. If the low volume sound is noise, bleed or the attack and release of a sound, it gets cut off.

You can read more about the most important things you need to know about mixing, and how to setup the compressors properly depending on the task you want to realize in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.

Mixing : Volume controls part 2 - Compressors / limiters

Compressors / limiters were originally introduced into the studio to stop loud volume peaks from distorting or saturating.

Compression and limiting are volume functions; their main purpose is to turn the volume down. They turn down the volume when it gets too loud - that is, when it goes above a certain volume threshold.

When the volume is below the threshold, the compressor / limiter does nothing (unless broken or cheap). The difference between compressors and limiters is explained later in the section on "Ratio Settings"

Compressor / limiter Functions

Compressors / limiters have two main functions (and three other minor ones.) The first function is to get a better signal-to-noise ration, which means less tape hiss. The second function is to stabilize the image of the sound between the speakers, which means more presence.

Better Signal-to-noise Ratio : Less Hiss

Recording extremely dynamic sounds, with a wide variation from soft to loud, requires turning the volume down so that the loud sounds don't overload and cause distortion. Distortion is against the law. Get distortion go to jail. But when you turn the volume down, the soft portions of the sound barely move the needles on the tape player. And if the needles are hardly moving on the multitrack, you hear as much tape hiss as you do signal. This condition is known as bad signal-to-noise ration and sounds very similar to an ocean: "shhhhhhhhhhhhhh."

By using a compressor to turn down the volume when the signal gets too loud, you can then raise the overall volume above the tape noise. By turning down the peaks, you can record the signal hotter on tape. Then the softer portions are loud enough so that you don't hear the tape noise.

When recording digitally, there isn't much noise to worry about. However, if you record too softly the quality of the recording is not as good. You are actually recording at a lower bit rate.

Therefore, compressors / limiters are also good to use when recording digitally. In addition, limiters can be used to keep very quick sound peaks that you might not even hear from going into the red and distorting.

Stabilizing the image of Sounds : More presence

 

After years of using compression to get rid of hiss, people realized that the sounds often appeared more present when compressed. By evening out the volume peaks on a sound, a compressor/limiter stabilizes the image of the sound between the speakers. A sound naturally bounces up and down in volume, as shown by the bouncing VU meter. When a large number of sounds fluctuate naturally, their bouncing up and down can become extremely chaotic - similar to trying to watch a 24 VU meters at once. A compressor / limiter stabilizes or smoothes out, the movements of sounds that result from these moment-to-moment fluctuations in volume. Once compressed, the sound no longer bounces around much, so the mind can focus on it better. Therefore, the sound seems clearer and more present in a mix.

The busier the mix (the more instruments and the more notes per instrument), the more sounds in the mix are normally compressed. This is because the more sounds and notes, the more chaos. It is difficult to keep track of a number of instruments in a busy mix in the first place. By stabilizing the sounds, the entire mix becomes clearer. Most "acoustic" sounds are compressed, although different engineers disagree about whether to compress the drums or not. Often the kick drum is compressed, and then the entire drum set is compressed with an stereo compressor.

Once the sound has been stabilized, you can then turn up the overall volume and put the whole sound right in your face. This is commonly done in radio and TV commercials to make them sound louder, so that they jump out and grab your attention. This might be annoying in radio and TV commercials, but it's great for a lead guitar or any other instrument you want extremely present in the mix.

This also works when putting sounds in the background. The problem with low volume sounds is that they can easily be lost (masked by other sounds) in the mix, especially if the volume of the sound fluctuates very much. Therefore, it is common to seriously stabilize sounds that are going to be placed low in the mix with compression. They can then be placed extremely low in a mix without fear of losing them.

A better signal-to-noise ratio is obtained by compressing the signals on its way to the multitrack. However many engineers will also compress the signal on its way back from the multitrack during mix down to stabilize the sound even more.

Sharper or Slower Attack

Besides less hiss and more presence, a compresor / limiter also makes the attack of a sound sharper. Once you turn down the louder part of a signal, a sound reaches its maximum volume much quicker.

With a shorter and sharper attack, sounds are much tighter, punchier, more distinct, and more precise, which makes them easier to dance to. On the other hand, a higher quality, fast compressor will actually hep to remove sharp "spikes" on the attack of a sound - softening the sound. A good compressor can mellow out the sound of a guitar with a sharp edge on the attack.

More Sustain

A compressor / limiter is also used to create more "sustain." This is commonly used on a guitar sound. Just as a compressor is used to turn down the volume peaks to raise a sound above the tape noise, it can also be used to turn down the louder parts of a guitar sound, so the guitar can be raised above the rest of the mix. Sustain is also specially helpful for obtaining the desired screaming feedback (when the guitar is held directly in front of a guitar amp).

Compressors are sometimes used in the same way to create more sustain on tom and cymbal sounds. The sounds seem to last longer before they fade out or are absorbed into the mix. The tradeoff is that compressing toms and cymbals will bring their level down, so that you actually hear the bleed more. However, depending on your musical values and the project you're working on, you may want to give this a try.

Less resonance

A final solution of a compressor / limiter is that it evens out resonances in a sound. Resonances occur in two places in instruments: hollow spaces and materials. When a hollow space (like the body of an acoustic guitar) has two parallel walls, it will boost the volume of particular resonante frequencies. Tap on the body of an acoustic guitar in different places, and you can hear the resonant frequencies. Materials (like the neck of a bass guitar) will also resonate at certain frequencies, boosting the volume of those frequencies. Play any guitar and you will notice that certain notes are louder and more resonant than others because they are activating the resonances in the body of the instrument.

A compressor / limiter evens out the volume of these resonances by turning down the loudest part of a sound, which just happens to be resonances.

You can read more about the most important things you need to know about mixing, and how to setup the compressors properly depending on the task you want to realize in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.

Mixing : Volume controls part 1 - Faders

Volume faders control the volume of each sound in the mix, including effects.

David Gibson comes up with a very useful and detailed description about what all the studio equipment does and how does it affect our sounds in his book "The art of Mixing Second Edition"

Faders

Volume faders control the volume of each sound in the mix, including effects.

The level set for each sound is based on it's relationship to the rest of the tracks in the mix. When volume is mapped out as a function of front to back, you can place any sound or effect up front, in the background or anywhere in between by using the faders.

However, the level that you set a sound in the mix is not based solely on the Vader. If the level of the faders was the only thing that affected the volume of a sound in a mix, you could mix without even listening. You could simply look at where the faders are set on the console. There is more to it than that.

When you set volume relationships in a mix, you use apparent volumes to decide on the relative balance - not just the voltage of the signal going through the fader. The apparent volume of a sound in a mix is based on two main things, fader levels and waveform, and another minor one, the Fletcher / Munson Curve (See the "Fletcher/Munson Curve section below). First, the level of the fader does affect the volume of the sound. Change the level of the fader and the sound gets louder or softer.

Fader Level

When you raise a fader on a mixing board, you are raising the voltage of the signal being sent to the amp, which sends more power to the speakers, which increases the sound pressure level (SPL) in the air that your ears hear. Therefore, when you turn up a fader, of course the sound does get louder.

The decibel (dB) is used to measure the amplitude of the signal at each stage of this circuit. In fact, there are very specific relationships between voltage, wattage, and sound pressure level. Decibels are the main variables that affect the apparent volume of a sound. However, there is another important factor: The waveform of the sound.

Waveform (or Harmonic Structure)

The waveform, or harmonic structure, of a sound can make a big difference as to how loud we perceive a sound to be. For example a chainsaw will sound louder than a flute, even if they are exactly at the same level on the VU meters. This is because the chainsaw has harmonics in the sound that are irritating - or exciting, depending on your perspective. These odd harmonics are dissonant to our psyche, which makes them seem louder to us. Therefore, a sound, even if they are at the exact same volume in the mix. A minor factor contributing to the apparent volume of a sound is the Fletcher / Munson Curve.

The Fletcher / Munson Curve

The biggest problem with the human hearing process is that we don't hear all frequencies at the same volume - specially those at low volumes. (Fletcher and Munson did a study that shows just how screwed up our ears are.) This is why there are loudness buttons on stereos - to boost the lows and highs. You are supposed to turn them on while listening at low volumes. However, most people like extra lows and highs, so they leave the switch on all the time.  The main point here is that you should check your mixes at all volumes all of the time because you won't be hearing ass and tremble as much as you should. Also, whenever you do a fade at the end of a song, the bass and treble will drop out first. Technically, your ears give you the flattest frequency response at around 85 decibels.

Apparent volume is, therefore, a combination of decibel level, waveform, and the Fletcher / Munson Curve. But relax. Your brain has all it figured out. Most people has it all figured out. Most people have no trouble telling whether one sound is louder than another (although most of us need to learn to hear smaller and smaller decibel differences). Your brain quickly calculates all of the parameters and comes up with the apparent volume. All you have to do is listen to the overall apparent energy coming from each sound in the mix. It is apparent volume that you use to set volume relationships in the mix. You don't look at the faders; you listen for the relative volumes.

You can read more about the most important things you need to know about mixing, in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.

Mixing : Functions of Studio equipment according to David Gibson

Get the most of your studio equipment knowing the functions of all of the different devices

Dave Gibson brings a comprehensive guide to understanding functions of studio equipment in his book "The art of Mixing" according to him studio equipment can be broken down in the following categories.

To simplify all of the functions of a huge variety of studio equipment, I have broken down all of the functions of studio equipment into categories based on the function of each piece in the recording studio.

1. Sound Creators : all instruments, accosting to electric, voice to synths
2. Sound Routers : mixing boards, patch bays, splitters
3. Storage Devices : recorders, tape players, sequencers, samplers
4. Sound transducers : miss, pickups, headphones speakers
5. Sound manipulators : processing and effects.

Effects rack

Sound creators range from acoustic to electric instruments, from voice to synthesizers.

Sound creators

Sound routers route sound from one place to another. Mixing boards route the signal to four places: the multitrack, the monitor speakers, cue headphones (for the band out in the studio), and the effects (so we can have a good time). Patch bays are just the back of effects - the back of the mic panels, the back of the multitrack (inputs / outputs), the back of the console (ins / outs) and the back of the effects (ins / outs) - located next to each other so we can use short cables to connect them.

Sound routers

Storage devices store sound or MIDI information and play it back. Tape players store digital or analog sound; sequencers store MIDI information. Some storage devices can be used to edit the sound while it is stored.

Storage devices

Sound transducers take one form of energy and change it into another. Microphones take mechanical energy, or sound waves and change it into electrical energy. Speakers take electrical energy and change it into mechanical energy, or sound waves. Likewise pickups on guitars take the movement of the strings and change it to electrical energy.

Sound transducers

You can read more about the functions of studio equipment and the most important things you need to know about mixing, in Dave Gibson's book "The Art of Mixing 2nd Edition" A visual guide to recording engineer and production.