Sound Design COMPLETE course - EVERYTHING you need to know to craft any sound.

hi there if you're interested in sound design and learning how to create your own sound textures you may have fallen in the right place i'm planning to do a series of videos that will be called sound design theory and this particular video will serve as an introduction for it i did a whole series on music theory before and now i would like to do something similar with sound design my purpose here is not to give you a recipe for a particular sound in each video or at least that's not what i will begin with instead i would like to go deeper and explore each tool we have and explain exactly how they will affect the sound and then we'll see several ways to use each of those tools i will not say something like then other tape saturation will sound good instead i will explore with you how saturation process the sound and show you different examples of what can be achieved with it so you can choose better tape saturation for example is what you need i think that when you understand really how an effect or a module works it opens a lot more doors and will be easier for you to recreate a particular sound or to create your own or at least that's what i believe in so the plan is to see different types of sound synthesis and then explore all the components that make a synth like an oscillator a filter envelope etc and then we will explore a lot of effects that you can also use on samples like saturation delay flanger etc and then we could move on to some more particular techniques and maybe do some sound exploration where we build a particular sound together either for musical use or for sound effects like you could use in a video game for example so if you want to start now i think my old episode about harmonix is a good introduction to this series as i talk about the basis of sound synthesis and i might not repeat all i've said there so i'd recommend watching it first if you haven't already i will add it to my sound design theory playlist anyway so it would be easier to find for each episode i will do my best to describe exactly how each module we see processes the sound which parameter you have at your disposal to manipulate them and some examples on how you can use them so hopefully you will have enough to start designing your own sounds right away to do that i plan to use examples on three main supports ableton live and is built-in plugins which is my door of choice vcv rack which is a free modular synth simulation if you don't know what a modular synth is i did a quick explanation at the beginning of my review for the volca modular and i'll do some examples on serum as well which is one of the most popular vst out there as the episodes might be quite long i might shave off the intro and the outro on each of them to have a straight to the point format which is also why i wanted to make this separate video and because these episodes might be quite dense as well in the description you will find the timings for each part of each video so if it is difficult to watch it in one sitting it would be easier to watch it one chapter at a time it would also make it easier if you want to get back to it to get a particular info so i'm basically trying to create a video encyclopedia on sound design so let's begin i hope you enjoy it truth is in nature there's no sound that vibrates on only one frequency at a time a sun that vibrator only one frequency is a sine wave that's what only one frequency sounds like any other sound is in fact always a cocktail of a lot of frequencies among them the lower frequencies is called the fundamentals that's often the note we hear and the one we can sing and all the other frequencies are called the overtones we split these other tones into two families the harmonic overtones and the inharmonic overtones the harmonic overtones participate to the musical note we hear in the sound the one that we can sing they are parts of the harmonic series of the fundamental which means that their frequency is multiple of the fundamentals frequency the fundamental being the first harmonic the second harmonic being twice this frequency which is the octave the third harmonic being three times this frequency which is the fifth the fourth harmonic is another octave and the fifth harmonic at the third etc they all participate to the musical notes we hear in the sound in fact when a string of a guitar vibrates all these harmonics are present in the sound we hear when you play a natural harmonic you isolate some of these frequencies that were already there with an open string we hear the fundamental with your finger touching the string just above the 12th fret which is the middle of the string you hear the second harmonic with your finger touching the string above the seventh fret which is the third of the string length you hear the third harmonic etc all the other overtones that are not part of this harmonic series are called inharmonic overtones and participates to the tone of the sound if a sound have too much in harmonic overtones then we can't really hear a note in it in this case we talk about inharmonic sounds it's the case of a percussive sound like a snare symbol for example drummers know that you still have to tune your drum kits and this is because the sounds found in nature are rarely 100 harmonic or inharmonic but more from the mixture between the two so these harmonic overtones participate to the musical note we hear in the sound and these inharmonic overtones affect the tone of the sounds as a practical example a pure sine wave is the only sound that have no overtone because that's how one frequency sounds then if we add odd overtone from our harmonic series the fundamentals begin 1 so we add the 3 5 and so on then our sine wave begins to become a square wave then if we begin to add the other overtones of our series our wave turns into a sawtooth wave this means that the sawtooth wave is richer in overtones than a square wave which is richer in overtone than a sine wave but the three waves are 100 harmonic sounds and are actually at the basis of some design in any synthesizer as an example of a hundred percent in harmonic sound you have the white nose which is all the frequencies between 20 hertz and 20 kilohertz all at the same level which is at the basis of a synthetic snare or symbol sound mix harmonic sounds with inharmonic frequencies and you can theoretically recreate any sound you've heard in fact overtones are the reason why your piano and the trumpet sounds different is because they don't have the same overtones in it harmonic and inharmonic and each overtone resonate at different levels let me show you [Music] we can clearly see the harmonic series in the three first sounds the lower sun being the fundamental and with harmonics at different levels and we can also see the inharmonic component of each sound in the two last sounds we can't really see any harmonic series these are inharmonic sounds to create your own sound from scratch either for musical use or to design sound effects you will use three families of elements a sound source would it be an audio sample or an oscillator with a basic waveform will give the bass sound you're gonna work with it will then be processed by different effects it could be filters distortion or anything else and then you're gonna have what i call control modules such as lfos or envelopes they don't make sound on their own but they can control parameters of the sound source and the effects to bring movement in the sound in this video i will focus on the sound source component and because audio samples can be basically anything we'll focus on the oscillator first to keep it simple and to get the core principles down but a lot of what we can do with them can be done with audio samples as well vco stands for voltage control oscillator and is the heart of any synthesizer for example here on serum there are two of them and here on ableton's operator there are four the sound they provide depends on the shape of their oscillation their waveform we talked about these in my old video about harmonics and overtones i will leave a link in the description for that but to sum it up the most common waveform are the sine wave the triangle wave the sawtooth and the square wave they all sound different and learning to recognize them will help you a lot in your sound design the sine wave has a very smooth and soft sound it is often this type of waveform that is used as a sub bass to double other synth sounds because it is made of only one harmonic which means that a pure sine wave is made of only one pure frequency the square wave has a more hollow sound without any effect it reminds the tone of old video games and it also have more harmonics than the sine wave so it also sounds richer with a square wave you will often have a parameter called pulse width this would make one part of the wave thinner as the other part gets wider the more asymmetric the wave the more metallic and nasal the sound it also makes it sound a bit more like a sawtooth wave [Music] [Applause] nowadays you often have this pulse width parameter for other waveforms too like in serum you can find it here where it says pwm for pulse width modulation with all these morphing options these are a lot to try already the triangle wave is halfway between the sign and the square both in softness and in harmonic contents it's pretty soft [Music] and the sawtooth waveform sounds more aggressive it is also richer in harmonics than the others so it sounds fuller and the fact that it sounds fuller makes it a good candidate to work with filters [Music] so if you want to make a smooth sound like a flute you might want to choose a sine wave or a triangle waveform and if you want to make a more aggressive sound like big horn stabs or an aggressive bass sound you might want to go for a sawtooth wave in the context of a synth you can obviously control the pitch of the oscillator with the keyboard each key given a different note but in the context of a modular synth like vcv rack an oscillator on its own doesn't have much control if it's not linked to anything it will only output a continuous buzz even to control its volume you need a separate module called vca for voltage control amplifier here for example in this setup i have this module that is simply my audio output so we can actually hear the sound the sound is provided by this vco the oscillator that goes into a vca the amplifier and straight to the output the vca acts like a volume knob and on the oscillator i can change the frequency this one is for the tuning so the frequency as well but with fine tuning and the three others do nothing at the moment [Music] but even though oscillators are very basic on their own you can already do several things with them you can stack several oscillators with different waveforms to get new tones so there are a lot of combinations to try already you can try with two or three or more different oscillators and also try to change the volume of each one as it can drastically change the tone [Applause] if you use several oscillators with the same waveform you can separate them by different intervals to make the sound richer usually an oscillator is doubled by an octave or a fifth it doesn't seem much like that but when you play a melody with it your brain will hear it like just one instrument and it can help the sound go through a mix if it's double higher or it can have more weight if you double it an octave lower which is usually done with a sine wave or a triangle waveform [Music] [Music] [Music] or if you use two oscillators with the same waveform you can also detune them slightly usually an oscillator would be detuned down slightly and the other will be determined up so they create kind of a phasing effect that can thicken the sound and also add some motion due to the phase of the oscillation going in and out of sync it creates a waving sound kind of a beating in the sound the more you dig in the oscillator the quicker the beating so [Applause] this beating appears because of phase cancellation when two sound waves play together they'll be added up so where the two waves have positive values they will sound louder and where they have opposite values they will cancel each other out so here because the two oscillators are slightly detuned they go slowly in and out of sync and that's the beating we're hearing so when you do that with sine waves or triangle waves the beating will be even more drastic because they are not made of a lot of harmonics and if you do that with square waves or so tooth wave the beating will be a bit more shallow because they have more harmonics more frequencies to cover up the hole in the sound this detail trick is implemented in serum in a very handy way here you can choose how many voices you want and with this detune knob you can set how much you want to detune them and you can also reduce the volume of the detune sound to mix it with the original sound now technically if you want to play several notes at the time to make chords you will need one oscillator per voice so that would be three oscillators to play a chord of three notes for example but software synth and even some hardware synth directly give you the option to play either polyphonic or monophonic while polyphonic allows you to play chords monophonic also have some advantages monophonic can sound cleaner as there won't be any notes overlapping when you don't want them to but it also allows you to use some glide also called portamento basically it will make the pitch glide smoothly from one note to the next the more potamento you add the longer the glide time and that's a very neat feature that can really transform the feel of your sound it's very cool to make sequences with some notes that are linked and other they are detached [Music] it's also worth noting that these oscillators are not the only sound source you can have in the synth you can also have white noise which sounds like a static noise it is a sound where all the frequencies between 20 hertz and 20 kilohertz are present it doesn't sound that appending on its own but it can be very useful in different situations like to fill the background of a mix for instance [Music] it is a good basis to make synthetic hats or snares for example and as all the frequencies are present it is also a good candidate to work with filters you can also have pink noise which is similar to white noise except that the higher frequencies are a bit quieter so it sounds smoother to the ear more recent synths like serum also use wavetables which allows you to have several waveforms in one oscillator they have a parameter to move the waveform from one shape to the other smoothly that can create some pretty cool sound that almost sound alive if you automate it some synthesizers like serum even allow you to draw your own wave shapes or to import them from an audio file which already unlocks a ton of possibilities with all the techniques mentioned above talking about audio file your sound source can also be assembler which will use an imported sound directly so you can play note with it like an oscillator so that could be anything a drum sound a live instrument like a guitar or a back pipe anything you want [Music] right off the bat there are a lot of things you can do with only oscillators and samplers but everything i showed you was all by stacking oscillators upon each other even though layering is a very important concept in sound design there is so much you can do with two oscillators to create your own sound so today we are going to see that and talk about sound synthesis there are a lot of types of sound synthesis so let's start right away with the first one additive synthesis is creating sound by adding harmonics in the sound that can be achieved by stacking oscillators on top of each other just like we did in the last video but most of the time we will use sine waves to do that so we can add harmonics one by one each harmonic you add will change the waveform because the waveform is dependent of the harmonic content and vice versa so changing one will change the other you can do additive synthesis in ableton's operator for example here in this area every vertical bar represents a harmonic of the harmonic series and the height represent their volume if you don't know what the harmonic theory is it might be a good time to watch my old episode about harmonics and overtones that i will link in the description everything is explained there so if you have only the first harmonic the fundamental you will have a sine wave then if you add a harmonic every two you would have more of a square waveform and if you add every harmonic you would begin to have more of a sawtooth waveform what's interesting in additive synthesis is that you can control the volume of each harmonic independently so you could shape the sound with more flexibility [Music] you can also do additive synthesis in serum by clicking this pencil icon in the oscillator section and in this window you will have the same display with all these bars with a visual representation of the waveform you are creating [Music] so [Music] so [Applause] subtractive synthesis is starting with a sound that is rich in harmonics and then removing some of them using filters there are a lot of different filters to choose from and we'll see what they can do in a dedicated video but for now you can remember that subtractive synthesis is gradually removing harmonics using those filters to give you an example regardless a low pass filter will cut the high frequencies of a sound so it can make it less harsh or more muffled [Music] and a high pass filter will cut the low frequencies of a sound which can be useful to leave some space for other layers in your sound for example [Music] so subtractive synthesis works particularly well when you start with a sound with a lot of harmonics to give you an example of subtractive synthesis in its purest form you can start with weight noise which contains all the frequencies and then use filters or an eq to let through only the frequencies you want [Music] in a wavetable synth like serum or ableton's wavetables the oscillator is called a table and it actually contains several waveforms you would then have a knob to cycle through all of them and morph your sound this can be a good way to add motion to your sound to make it more organic varying the waveform in conjunction with the opening of a filter or the drive of a distortion can really make your sound come alive this is interesting because you could have any way for morphing to any waveform so the possibilities are endless [Music] some of this synth-like serum even allows you to draw your own waveforms or you can import an audio sample and it will extract the waveforms from it for example if i take this sound and i import it in serum it would give me all these waveforms that i can use in my oscillator [Music] among these waveforms you can do a selection you can erase some of them you can keep just some of them and if you have just a few different waveforms loaded here you can actually like interpolate waveforms between them so you can move smoothly between them to do that it's in the morph section let's try more spectral for example see the first one is in index number one and the second one is in index 256 which means that there are actually 255 waveforms to transition from the one to the other [Music] am stands for amplitude modulation and the amplitude of the signal is basically the volume of the sound amplitude modulation means that the volume is modulated the volume changes following another signal so for am synthesis you will actually need two oscillators one is the sound source that provide the sound you can hear it is called the carrier and the other is the one controlling the volume knob of the first that is called the modulator so for example we have an oscillator with a sine waveform that is our base sound the carrier and then we have an lfo which stands for low frequency oscillator that will provide also a sine wave but very slow if we make this lfo control the volume knob of our first oscillator it becomes the modulator at this rate you can hear the sound go louder than quieter along with the lfos oscillations for now it's no big deal but if the lfo then goes faster and faster it starts distorting the sound when the modulator oscillates fast enough it begins to distort the shape of the waveform itself and it begins to add harmonic content to the original signal [Laughter] it will add several harmonics for each harmonics already present in the original sounds so if i add another sine wave in the carrier so we begin with two harmonics you can see a series of harmonics being added for both changing the amplitude or volume of the modulator changes the overall volume of the added harmonics from there you can experiment with different waveforms for the carrier and for the modulator [Music] [Music] it's worth noting that you can replace this lfo for a simple oscillator as well an oscillator and an lfo are basically the same thing only the lfo slates way slower one last thing about am synthesis have you heard about ring modulators these are devices that can make a sound more metallic that sounds a bit like a bear the name ring modulation comes from the fact that early analog devices that made rin modulation used four diodes linked together in the shape of a ring well ring modulation is essentially the same thing that amplitude modulation the only difference is that with ring modulation only the added harmonics remains and the original frequency of the carrier disappears whereas with amplitude modulation both added harmonics and the original frequencies of the carrier remain apart from that they work in the same way fm stands for frequency modulation and frequency is basically the pitch of your oscillator so fm synthesis is similar to am synthesis in the way that they both involve a carrier and a modulator but here with fm the modulator will control the pitch of the carrier in the same example than before i will link the lfo to the frequency of the oscillator [Music] you can hear the pitch going up and down along with the lfo and in the same way when the oscillator rate of the lfo goes up it starts distorting the sound of the carrier but it adds a lot more harmonics creating potentially a much richer sound one of the differences with am synthesis is that here with fm when you move the amplitude or volume of the modulator it also affects the harmonic content that is added to the sound not only the overall volume but also the distribution of the harmonics [Applause] [Music] [Music] now this is the core principle of fm synthesis it can go a lot deeper for example we can add a third oscillator before this modulator one so its frequency goes up and down here is how it sounds at a slow rate so if i crank up the frequency of this oscillator we can get yet another sound from there you can add other oscillators and connect them in many different ways i'm showing you this because fm synthesis was popularized by the dx7 it was a synth that used the oscillators that you could connect in many different ways and this architecture was so popular and versatile that it's been copied in many synths like in the volca fm for example other software decks which you can download for free you can do many different sounds with fm synthesis it's really a rabbit hole that goes a very long way so here are a couple of examples from dext library [Music] i will also add wave shaping synthesis which generally starts with a simple waveform such as a sine wave a triangular wave or sawtooth wave that is then distorted by a wave shaper the wave shaper will basically change the shape of the incoming waveform based on a function and there are a lot of things you can do with it it is a type of distortion so we'll see that more in detail in the episode about distortion and saturation but just to name drop what you can do with it you can do some saturation asymmetric saturation wave folding or phase inversion for example heart sync is not really considered as a type of synthesis i don't think anybody talks about heart sync synthesis but it creates a very particular sound so i wanted to include it here as well it is close to am and fm synthesis in the sense that it requires two oscillators one controlling a parameter of the other and what happened is that every time one oscillator finishes a wave cycle it resets the cycle of the other oscillator that creates a break in the waveform and it creates a distinctive distortion but having one oscillator resetting the phase of the other means several things one the two oscillators will have the same base frequency because the master oscillator will force the slave oscillators period to reset at the master's frequency so if you change the master oscillator's frequency it will change the pitch and if you change the slave oscillator frequency it will change the timbre [Music] two you will hear the effect better if the two oscillators are detuned or played two different notes because if they are perfectly in tune and they play the same note then their phases would be perfectly in sync and it won't cause any break in the waveform it would be seamless and three the waveform of the master oscillator doesn't really matter because it is only its frequency that will determine when to reset the period of the slave oscillator usually the frequency of the master oscillator is the frequencies of the notes you play on the keyboard and the frequency of the slave oscillator is either tuned from these nodes or it can be set to a constant frequency this really depends on how your synth is built and the options you have on it last thing to keep in mind with hardsync is if the slave oscillator is tuned to a lower frequency than the masters later it will be forced to repeat before it completes an entire cycle and if it is tuned to a higher frequency the slave oscillator's waveform will be retriggered after one or a few complete cycles and this can give the impression that two nodes are playing at the same time so if you sweep the tuning of the slave oscillator you will actually hear kind of a harmonic series [Music] granular synthesis doesn't use an oscillator as a sound source instead it uses an audio sample well it could still be a sample of a sound made with an oscillator but still it uses an audio sample and slices it into a lot of tiny audio bits generally between 1 millisecond and 50 milliseconds each these tiny snippets of sound are called grains these grains can be layered on top of each other played at different speeds different length different volume etc this kind of synthesis is handy to create mellow textures often referred as clouds [Music] but it can do much more there is a free max for life plugin that allows you to do granola synthesis in ableton it's called granulator 2. here you can place your sample let's take this one for example and there you can define the size of the grains where they are reading from move the playing heads while you play change the pitch of the grains randomize these parameters and more check out this plugin it's free and very fun to play with granular synthesis is also used by some built-in functions in ableton like the time stretch the time stretch works in a very similar way in other doors but i don't know them as much as i know ableton so i will use this one for the example so time stretch allows you for example to make a sound longer without changing its pitch go go to do so ableton will cut your sound in several grains and then repeat some of them to make the sound longer you can then change the size of this grain in the warp mode section of your sample by default it's on beats in this mode ableton will make a new grain with every transient it finds it's good for small stretches or if you want to make your sound shorter but for bigger stretchers you might want to try other modes first still in beat mode you can set the grain size to a subdivision of your tempo which is really nice to make a rhythmic texture in sync with your tempo or you can make the grain shorter by clicking on this little arrow to make sure each grain is red only once and by reducing this value it is very nice to make a loop sounds snappier for example [Laughter] you can also use the texture mode that allows you to set the size of the grain to tania bits all the complex and complex pro functions which are supposed to be of best fidelity to the original sound but with more extreme stretches it introduces some metallic terms that can be very nice also very last tip for this video to create your own waveform you can also load a sample in a sampler and select a very tiny part of it and play it in loop because this part is played in loop very quickly it will just serve as a new waveform i used to do that a lot to design bass sounds starting with a kick drum sample and these are the eight audio synthesis that i wanted to share with you today additive subtractive am fm wave table wave shaper hearthsting and granular synthesis that was a quick overview of how they work but we'll see them again in this series most of these types of audio synthesis add a lot of harmonic contents they do very rich sounds so it's often a good idea to combine them with filters to tame them a little filters and eqs are probably the most used effects in music production would it be for mixing and mastering sound designing or live performances filters do a very simple task really they cut some frequencies of the sound you're designing so they are at the core of subtractive synthesis they can either be used to remove unwanted frequencies from a sound or to leave more space for other layers or other tracks in a mix in this video we'll see a lot of different types of filter each allowing you to cut or lower frequencies in a different way so you can hear how they sound but just before we begin here are a couple of things that are important to mention because filters will remove frequencies from a sound they work better if you start with a sound that is rich in harmonics like a sound made with fm synthesis or with a lot of distortion for example filters won't work particularly well on the sine wave for example as the sine wave is only one frequency so that's not a lot of content to cut to shape the sound and in serum you can choose the type of filter you want to use here and then choose what you want it to filter oscillator a oscillator b or noise you also have another filter in the fx section and another extra filter in the distortion effect so filters generally come with two main parameters the cutoff which is the frequency where the filter starts affecting the sound and the resonance also called q which amplifies more or less the frequency around the cutoff point the low pass filter also called the high cut filter is probably the most common it will cut all the frequencies above the cutoff point keeping only the lowest part of the sound that generally makes the sound smoother and with the resonance you can control how much the frequencies are boosted at the cutoff point though this filter doesn't cut abruptly every frequencies above the cutoff point it cuts them progressively more and more the further you go from it so it cuts them with kind of a slope you'll see sometimes a number attached to the low pass filter like lp-12 lp18 or lp24 it refers to the stiffness of the slope the higher the number the more abrupt the cut for a low pass 12 for instance the 12 stands for 12 db per octave so frequency one octave above the cutoff point will be lowered by 12 db and a frequency two octaves above the cutoff point will be lowered by 24 db this gives you an idea of the slope of the filter so here are a couple of examples of low pass filters uh [Music] [Music] [Music] oh [Music] the high pass filter also called low cut filter will cut all the frequencies below the cutoff point and keep only the higher ones this usually makes the sound thinner and sometimes harsher it is useful to keep only the high end of the symbols for example or to keep only the texture of one sound to layer it with another sound similarly to the low pass filter you can sometimes see a number attached to the high pass filter it also refers to the abruptness of its loop [Music] the band pass filter cuts the frequencies above and below the cutoff point while boosting the frequencies around it the resonance here controls how much the frequencies are boosted a common way of using this filter is to make a telephone effect when it is put on a voice recording for example [Music] shhh [Music] [Applause] [Music] the notch filter also called band cut filter will cut a band of frequencies around the cutoff point the resonance here will set the width of this band a higher resonance will result in a narrower band being cut and a lower resonance will result in a wider band the notch filter is useful to get rid of a narrow slice of frequencies to remove a small part of the sound you don't want i personally like to move the cutoff point while a note is playing with the lfo for instance to add movement to the sound [Music] there are actually two types of comb filters they are fit forwards and there are feedback con filters let's see the fit forward form first the feed forward comb filter is like a combination of several notch filters it will be a succession of several points that will cut several bonds of frequencies they are sometime called flanger comp filters when the frequencies that occurred are multiples of its first frequency so this would be the first frequency this one would be at two times this frequency this one three times the first frequency and so on following the harmonic series and other for what comb filters are sometimes called also phaser comb filters but we'll see that when we'll talk about flanges and phases the cutoff knob would move the series of notches and the resonance will set the depth so these comb filters can be used to cut a frequency with its harmonics [Music] the feedback form of the comm filter will work in the opposite way it will cut all the frequency and let through only the frequencies around the cutoff points and its harmonics so it's like a series of narrow band pass filters this type of filter can be very cool to use on white nose to let a particular node go through for example to have a kind of breathy tone [Music] [Music] oh [Music] foreman filters are designed to mimic vowel sounds such as e a or o it does that with a series of band pass filters usually between two and five they let through only some frequencies selected depending on the vowel we want to reproduce additional point will refine the precision of the vowel you can see here a more precise chart giving the frequencies of five points for five different vowels for a e i o and u for five different types of voices from the lower to the highest bass tenor contour tenor alto and soprano if you want to see it more in detail you'll find it in the description for example let's try to make a formant filter with several band pass filters let's do it with three points i will have an oscillator with a sawtooth waveform because it's rich in harmonic and then i will make an effect rack with three channel with one filter on each channel this way the sun will go in parallel in each of this channel and will hear the three outputs at the same time each output giving one frequency from one filter so i will take three frequencies from the chart for a for eternal voice for example and i will copy each of them on a different bandpass filter so here is how it sounds like now we can also hear what it does with a different oscillator [Music] and there i did exactly the same thing with different values for different vowels to see how they sound [Music] the high shelf filter will boost all over all the frequencies above the cutoff points equally it is not that common as a standalone plugin but it is more often seen in equalizers but we'll see those in a minute the resonant here will change the steepness of the transition slope between the boosted part and the rest of the signal and then you'd have a gain knob to set the strength of the boost or the cut a high shelf filter can be nice to add a bit of brilliance or to reduce the harshness of a sound a bit if you don't want to cut the high frequencies completely the low shelf similarly to the high shelf will boost or lower all the frequencies below the cutoff point it's also pretty rare as a standalone but it's more frequent in the nicu [Music] [Music] [Music] a tilt filter also called tilt eq is like a combination of high shafts and low shelf filters it will either boost the trebles and lower the bases or boost the bases and lower the trebles the cutoff frequency act like a pivot point then you'd have a gain knob to set the strength of the effect and the resonance will set the size of the slope of the transition [Music] the all-pass filter is a bit particular because it has what we call a flat frequency response which means it doesn't cut or post any frequency instead it delays the signal a little by an amount relative to the frequencies it delays that means that the base frequencies can be delayed differently than the higher frequencies they are not really used as standalone filters but they are used inside phaser effects so i just wanted to let you know that they exist we'll talk more about them when we talk about phases parametric eqs are like a combination of several filters it's represented with a screen representing the frequency spectrum with the lower frequencies on the left and the higher ones on the right and there's a line on which you can add points each point you add is a new filter and you can choose if it acts like a low pass or high pass a high or low shaft or a point eq each point will then have three settings the frequency which is similar to the cutoff point of the filter it determines where the point is on the frequency spectrum the gain which determines how much the frequency around the point are lowered or boosted and the resonance of a noted q which determines how wide or narrow is the frequency range affected by that point it is one of the most versatile and most used effect in both the sound design phase and the mixing phase to make every sound of the track sound well together a good rule of thumb is to lower frequencies more than you boost them it is often advised to cut the frequencies you don't want more than boosting the ones you like another good rule of thumb is to use a high gain value with a high resonance and a little gain value with a small resonance so if you want to cut a frequency drastically it's often better to use a higher resonance with a narrow range of action and if you use a small resonance value to act on a wider range it's often better to be subtle with a gain knob with this in mind an equalizer is also an excellent tool to dissect a sound to see what it's made of when you use an eq you can take a point with a rather high resonance and a high gain and sweep it across the frequency range it is like a magnifier to hear each part of the sound if some region sounds too harsh too loud or too dissonant you can lower them step by step it's a good way to really carve your sound some eq even allow you to hear only what is affected by the point you're moving on ableton's eq8 is this button here some eqs also allow you to process differently the center and the sides using a high pass filter on the sides is a good way to ensure the base part of your sound stays in mono which is often what you want and some eqs also allow you to process differently the left channel and the right channel which is always good to enhance the stereo wideness of your sound graphic eq are another type of eq where the frequency spectrum is sliced into several bands you would then have a slider on each one to control how much you want to boost or cut the frequencies in that band the bands are already set so you don't have control over the frequency of each point nor over the resonance of each point which is often fixed but this kind of eq often allow you to have a lot more fun to play with lfos and envelopes are two control modules they won't make any sound by themselves but they are used to control other parameters and they can do amazing things they are very common and powerful modules to the point that you can find them on almost every synthesizers and you can even find several of each on a lot of them basically lfos and envelopes move parameters over time which brings movement to your sound and movement in the sound is a key aspect of interesting sound design well apart if you really want to go for a static aesthetic but you get what i mean so what do they do we saw the lfo briefly in a previous video lfo means low frequency oscillator so it is basically an oscillator but as the name states it oscillates very slowly most of the time below the hearable spectrum lfos come with different parameters the shape of the oscillation that can be a sign a triangle a square a sawtooth etc serums lfo even allow you to make your own shapes the rate which defines the speed of the oscillation you can then link that lfo to any parameter and the lfo will move this parameter along with its oscillation so you can achieve different effects with that you can link that to the amplitude or the volume of an oscillator to create a tremolo effect then if you crank up the rate of the lfo it begins to add harmonics to the sound and you enter in the domain of am synthesis you could also link the lfo to the pitch of your oscillators to create a vibrato effect if you crank up the rate and the amplitude of the lfo it also adds all the harmonics to the sound and you enter the domain of fm synthesis lfos are also commonly used on a filters cutoff with a low pass filter for example which will cut the higher frequencies you could create some wobble effect used a lot in dubstep and you can then automate the rate of the lfo to create rhythms [Music] on ableton's operator you can link the lfo to the amp or the picture of any oscillator by clicking on the lfo section and then selecting what you want to affect it to in this middle area and in serum you can find the lfo here you can then drag and drop the lfo on any parameter you'd like it to effect [Music] to create some movement in the sound and lfo is also often used on the waveform or on the pulse width of an oscillator so the source sound of the synth will change a little over time [Music] these are just a few examples to scratch the surface of what an lfo can do but remember you could use an nf1 pretty much any parameter of any module we've seen and will see in the future you can use an lfo sync to the tempo to use it rhythmically or use it with a slow rate to add just a little bit of motion [Music] or when you crank up the rate of the lfo you might discover new types of distortions just like when we modulate the pitch of an oscillator very fast to have fm you could find interesting things if you modulate very fast a parameter of another effect here is for example what happens when you modulate a phasor very fast [Music] try to use another one many parameters changing the wave shape the rate and the amount to make different effects some leftovers also allow you to use them in one shot mode in this mode the lfo shape would be red on it once which turns the lfo into an envelope essentially envelopes are also modules that won't make any sounds on their own their role is also to affect other parameters to see exactly what it does let's use an envelope linked to the volume of an oscillator that's the most common way to use it then it's called the amp envelope every synth would have a built-in envelope linked to the amp of the oscillator an envelope comes with four parameters ads and or for attack decay sustain and release these refer to different stages of what happens when you press a key of your keyboard to play a sound the attack determines the time the note takes to reach full power from the moment the note is played so with an attack at the minimum at 0 millisecond the note appears directly at its full power and with a longer attack the sound fades in the decay is the time it takes between the attack and the sustain and the sustain is the volume of the note when the key is held down if the sustain is all the way up the dk would have no effect because it would make a transition from full power to full power and the release is the time it takes for the note to die off after the key is released so the longer the release the longer the fade out so for example if you want to make a more percussive sound like axilla phone you can put the sustain and the attack all the way down so the length of the note will be controlled by the decay and you can then have a longer release so in case you play a very short note it will still resonate a little and if you want to make a pad sound that would be softer like a violin you can have a longer attack with a longer release you can have the sustain all the way up so you don't have to worry about the decay [Music] as you can see the envelope is pretty powerful to shape the sound as you want and i think it's one of the most important parts of a sound design but an envelope can control way more than the amplitude of an oscillator you can also control the cutoff of a filter which is also a very common thing to do it is then called a filter envelope when you see two envelopes on a hardware synth most of the time one is for the amp and the other is for the filter a filter envelope should come with an amount or a depth parameter that allows you to control how much the envelope moves the cutoff frequency this amount knob could go to positive as well as negative values to move this cutoff frequency upward or downward you can find this knob here on this module here in ableton after you click in the filter section and here in serum it's the little icon next to the knob you assigned the envelope to so when a note is triggered the cutoff starts where it is set on the filter then it moves up or down during the attack phase by an amount set by the envelope's amount then the cutoff will go back down or up during the decay phase to read the sustain value and when the node stops playing the cutoff goes back to the original value during the release phase you can then play with the resonance of the filter to hear more or less where the filter is cutting for example we have a sawtooth wave that is filtered by a low-pass filter and we have a filter envelope the envelope is set with a very short attack a little bit of decay no sustain and a shorter release so it would make like a percussive sound with the amount knob you can tell the filter envelope to bring the filters cut up upward by putting it on a positive value so there the cutoff will start quite low and will then be brought up by the envelope and then go back down with the decay and there we created a plucky bass sound that can go well with the kick to make like a trench track if we set the amount up to a negative value and turn up the filter cut off the filter sweep will then create kind of a huawei and if you want the huawei effect to open and close you can make it the other way around with a longer attack and a positive amount value for the envelope and if you want the wire to stay open when you play a note you can crank up the sustain then the release will set the time the huawei takes to close after another stop playing if you have an amp envelope and a filter envelope make sure the amp envelope let enough sound through if you want to hear the filter envelope you might not hear the filter's envelope decay if its attack is too short for example so make sure you have an actual sound going through so you can filter it an amp envelope and a filter envelope are super effective together to shape your sound but you could use envelopes on any other parameters so don't hesitate to try it on the pulse width or on the wave table position for example so to finish this video let's try to affect a couple of parameters with envelopes let's see what it does on the wavetable position for example it can really make the sound come alive [Music] [Music] no no no no no no no no [Music] since the beginning of this series we've talked about oscillators or geosynthesis filters envelopes and lfos and as much as i can't wait to talk about all the audio effects i still need to talk about one thing to wrap up this control section and that is midi controllers sequencers and applicators we saw how to shape the notes of a synth with oscillators and envelopes so now let's see how this synth actually plays those notes we'll see principles that are very useful in a modular setup like vcv rack but understanding them will be also very useful in other contexts to know what not to play and how to play them your synth needs four types of signal a gate that is either high or low basically it does the synth when a key is pressed when the key is held down the gate is high and when the key is released the gate goes down it is this info the envelope will use to know where to trigger its attack and release phases when the gate goes up it enters its attack phase followed by the decay followed by the sustain and whenever the gate goes down the envelope enters its release phase then there is the trigger it's a very short impulse that is sent every time a key is pressed if you play several notes one after the other the notes might overlap so the gate will stay open as there will always be at least one key that is pressed at all time but the trigger will send an impulse for every new key pressed so the envelope can reset for every note played [Music] you will see a lot of synth with an option called legato when legato is active the trigger will be ignored so the attack and dk phases will be played only for the first note and one envelope cycle can cover a whole musical phrase [Music] then there is the pitch because of course the synth will need to know what note to play in an analog setting the most common format is one volt per octave meaning the higher the voltage applied the higher the pitch of the note a difference of one volt between two nodes will result in a difference of one octave between them and then there is the velocity it is the strength with which each note is played it is generally linked to the amplitude of the note meaning that the harder you press a key the louder it is but it can be linked to any other parameter as well having these trims in four separated cables can be very useful in a modular setting so you could use the velocity to control the intensity of a distortion for example to have more distorted nodes when you play harder or if you have something generating triggers randomly you could plug it into an envelope generator so it would act like if a key was pressed every time it receives a trigger which is called to make self-generating patches you can also use the pitch to control the cutoff of the filter to open the filter more on higher notes this is actually called key follow and it can help you to have a more consistent tone throughout several octaves because if the filter is fixed low notes can have a lot of harmonics above the fundamental before reaching the filters cutoff and higher notes could simply disappear above this filter whereas if the cutoff moves with the node you could have the same number of harmonics above the fundamental every time resulting in more consistent sound but in the digital world the most used format to control a synthesizer is midi which gathers all this info in one cable if you use a midi controller to control your daw that's why you only have to plug in its usb lead which can also give power to your controller and that's very handy these midi controllers can come in many shapes like keyboards or pads but the principles stay the same each one will send this gate pitch and velocity data to control your synth and when you are drawing notes in the piano roll of your door you are essentially doing the same thing that you would do by playing a midi keyboard except you are programming it instead of playing it the thing that is super handy with midi is that you can write automations you can write or record the motion of any knob directly in your sequencer and you can also assign any number of knobs to a macro knob so when you move the macro knob it will move all at the same time and the two combined make a very powerful tool for example here in serum i have this macro knob linked to several things so if i move this knob it will move the cutoff of the filter the amplitude of the oscillator the position of the wave table and several other things which is very handy to create complex moving sound design [Music] in the same way midi controllers often have several knobs faders and wheels that you can set to control different parts of your synth now to control the notes yourself plays you don't necessarily need a keyboard pads or even to program the notes individually you can also use other modules like a step sequencer or an arpeggiator a step sequencer is a module that will send regular impulses to your synth so you can create a sequence of notes that will be played in loop by your synth for example in vcv rack you have the sec3 which is a 8 step sequencer meaning that you can make sequences of 8 notes with it and you can program three different sequences with one module they are represented by the three lines of knobs here each knob here represents the pitch of a note and the clock knob sets the speed at which the sequence is read so you can create your sequence on one of these rows and then you have the output for the pitch for each row there that you can connect to an oscillator and you have the output for the gate that you can connect to an envelope generator so it will be triggered every time and not displayed this sequencer doesn't have any output for the trigger because the gauge will stay open only for half the time of each step so the gate won't be overlapping in ableton live my favorite step sequencer is probably the max for life device mono sequencer just throw it on the midi track and you'll have a 16 step sequencer for the pitch of the notes that you can lock to a certain scale or leave it free of any scale [Music] and then you have different tabs on the left to sequence other things you have a sequence for the velocity to have a different velocity for each note the octaves if you want to play certain notes on different octaves the length of the note and the repeat sequence if you want to trigger several times the same note on a given step what's very cool about this sequencer is that every sequence is independent and you can make loops of different length for each of them so they would offset over time in a polyrhythmic fashion and that's perfect to create ever-evolving patterns basically an arpeggiator will take a chord as an input and break it down into an arpeggio playing one note at a time or you can also play just one note and the arpeggiator will repeat it at a certain rate let's take ableton lives our potato as an example you can see in the middle a rate knob that determines the speed of the arpeggio that can be synced to the tempo and next to that is the gate knob which determines how long the gate stays open for each note so a smaller gate means shorter notes on the top left corner is the heart of the arpeggiator this is where you can tell the arpeggiator how you want it to play the chords you give to it so there are several algorithms here to choose from up which will play the note from the lowest to the highest down which will play the notes from the highest to the lowest converge will play the notes on each extremity first followed by the notes in the middle to create this kind of pattern in addition to that you also have a chord trigger which will play the chord as is but repeatedly at a speed set by the rate knob and you also have several algorithms to make random patterns as you can see there are a lot of algorithms here to choose from but some arpeggiator also allow you to create your own sequence that will be transposed to every chord you play this rate gate and algorithm settings are the core of the arpeggiator and you should find these parameters on every applicator with more or less options for example on this one you can make it sensitive to the velocity of the note you play or you can transpose them to a particular scale or if you want to get creative with it you can unsync the rate and control this knob with an envelope or directly automate it to create an exponential rhythm [Applause] so you can use either a controller a step sequencer or an arpeggiator to control your synth and they all use gate trigger pitch and velocity information which you can then use to control other parameters for example i said before the velocity often controls the volume of the notes but you could also link it to the amount of the filter envelope so the envelope would open the filter more when you press the key harder one way to do this in serum would be by using the matrix tab there you can set the velocity to control an envelope amount or you could link the pitch of the note to the cutoff of your filter to do the key following i talked about earlier so the higher the note the more open the filter don't hesitate to try to control different parameters with the velocity the gate of the pitch this is an excellent way to make expressive patches or link several parameters to one knob and automate this knob in a rhythmic pattern that's also an excellent way to morph your sounds and combined with envelopes and lfos it is a very powerful tool to have [Music] now that we've talked a bit about sound sources and different control modules let's get to the real meat of this series and talk about audio effects today we'll start with distortion and saturation there are two different ways to approach the nature of a sound that are tightly tied together you can define the sound by the harmonics that are combined to build it describing what harmonics and inharmonics are in it and at which level that's the principle of additive synthesis for example where we build the sound by adding harmonics or you can define a sound by its waveform and that's what we'll be focusing on today the two are very much related as adding or removing harmonics will change the waveform and changing the waveform of an oscillator will change its harmonic content we talked a little bit about this in the second episode of music theory in five minutes if you haven't seen it already to carve your sound by affecting the waveform directly you can for example use a wavetable synth like serum that allows you to directly draw the shape of your oscillator just click on the pencil icon on the top right corner of an oscillator and it will take you to the editing window but that can be a little fiddly if you don't know where to start or where you want to go with it so another way to affect the shape of the waveform of an oscillator is to use a wave shaper here i am using the veggie wave shaper which is a free max for life module for ableton live a wave shaper is an effect that does exactly what the name let's suppose it takes an incoming signal and change its shape based on the function generally a wave shaper's function can be represented on a x-y graphic here the voltage of the incoming signal is represented on the x-axis and the voltage of the output signal is on the y-axis the shaping function is then represented in the graph in the center for example a graph that ramps up from -1 to 1 will leave the waveform and change an input of -1 will output a value of -1 an input of 0 will output 0 and an input of 1 will output a value of one a graph that runs down from one to minus one will invert the phase of the waveform as an input of minus one will output a value of one an input of zero will output zero and an input of one will output a value of minus one the graph being a straight line it won't change the shape of the waveform though these are very simple examples but with a wave shaper you can do many things for example you can emulate many types of saturation so to explore that further let's see how a saturation works basically when you boost the amplitude of a signal the sound gets louder but when the amplitude of the signal exceeds the maximal range the tip of the waveform gets clipped resulting in a distorted sound and this is what saturation is this can be replicated with a wave shaper with this kind of curve with hard corners where the curve meets the maximal and minimal values of the graph in this example every part of the incoming signal that is below minus 0.5 will be flattened at -1 and every part that is above 0.5 will be flattened at 1. you can then make the effect even more drastic by boosting the amplitude of the incoming signal or by bringing these two points closer to the middle making the slope in the center steeper to make the distortion more subtle and sound a little more analog we can get rid of these hard edges where the signal get clipped and replace it by a curve the smoother the curve the softer the saturation [Music] so there are two main parameters we can play with to make a saturation effect more or less extreme one is the size of the curve the wider the smoother and the tinier up to no curvature the more aggressive the sound and the other is how much we boost the signal beforehand which is often referred as the drive or gain on the saturation effect it can also be translated by the stiffness of the slope in the center the more vertical the more drastic the effect and all the terms used for different types of saturation or different combinations of these parameters from the smoothest to the more extreme there's the analog warmth or analog boost with a very smooth slope that barely distorts the waveform then the overdrive or tube that offers a gentle distortion often used in blues for example then the distortion with a harder curve that provides a much more distorted sound used a lot in rock and hard rock for example and above that are the first pedals that really squash the signal with a hard curve this is used a lot in hard rock and metal music for instance and at the end of the spectrum is the digital clipping with no slow pedal which gives a very harsh sound but the more extreme you go with the distortion the more harmonics are added to the sound which can work well for subtractive synthesis so harder distortions often go well in pair with filters [Music] this is also why saturation effects are often used with sub basses if the sub is too low for some speakers you can add harmonics to it so you can hear them better on smaller equipment full disclosure these are only the basic differences between all these types of saturation there are a lot of other things that can make several saturation effects sound different would they be digital effect or analog pedals some of the things have to do with how components like diodes would react in the circuit which can be emulated digitally but it's not in the scope of this video as it can get very specific and i am not an expert on that myself other differences have to do with the sampling rate which we'll see more in detail when we talk about bit crushers but one of the things i'd like to talk about is symmetry the types of distortion we've seen so far are symmetrical they distort the upper side of the signal in the same way than the lower side but you can distort the higher part differently than the lower part to make them asymmetrical which can give a different color to the distortion this is often used to distort the waveform very hardly on one side while having it very smooth on the other side this way you could achieve a very hard distortion that keeps a certain clarity or a certain warmth this type of distortion is often associated with tube distortion [Music] and there are two ways to achieve this asymmetrical distortion one is to have a curve that is different on the lower part but in the higher part and the other is to offset the incoming signal before it reaches the wave shaper or the distortion effect if the waveform is offset upward the upper part of the signal will be clipped before the lower part resulting in an asymmetric distortion let me show you one trick to do that in ableton live so you can try it with other saturation effects say you have an oscillator here that could be anything and it goes into a wave shaper the trick is to put this oscillator in an instrument track and create a second channel in this second chain put an operator and use only one oscillator with a fixed frequency of zero hertz so it won't oscillate at all it will just add a steady positive or negative voltage to the signal and to control how much of this voltage is added you can use the starting point parameter here so the oscillator will start at a different point of the waveform and stay there this way you can control how much you offset the signal defining how asymmetric the saturation will be [Music] [Music] [Music] this type of asymmetry is used a lot with first distortions so you can play with that and see if you find something cool so there we talked a lot about saturations and there's a lot you can do with this to model your sounds but that's still one facet of the wave shaper which can do much more and there is one last type of way shaping i would like to talk about and that is the sign fault function also often called wave folding visually on the wave shaper the sinefall function looks like this it's like a sine wave that goes back and forth between the top and the bottom of the graph what it does is that instead of clipping the top and the bottom of the waveform when it hits the maximum threshold it folds it back on itself hence the name with folding it is a type of distortion that can create a very harsh metallic tone by adding a ton of harmonics to the sound because it adds so much harmonic content it can make it a good candidate to use as a starting point for subtractive synthesis you can use it with a low pass filter that moves along with an lfo to create a dubstep wobble for example and you can make the sound even harsher by reducing the slope at the top and the bottom of the sine shape to make it look like a triangular wave and the more back and forth the curve does on the wave shaper's graph the more times you can fall back the incoming signal before saturating the effect a similar waveform transformation you can do is having a curve that looks like this with this type of curve the bottom half of the waveform will stay unchanged but the top half will be flipped on its head this type of distortion is often called rectify in some synthetic plugins [Music] now something i like to try is to add several distortion effects in the chain it works better if the distortions are different layering two different distortions is something you'll see a lot in the design of heavy guitar tones for example the idea is that because every distortion will add harmonics i would put a low pass filter after each one so each distortion will distort the sound a little then i filter out the higher harmonics then i distort it a little further rinse and repeat here is an example of a sound i made with ableton saturation in its wave shaper mode [Music] with all this you can design your own distortion effects creating different saturations and waveforms with different curves in a wave shaper or using presets you have at your disposal don't hesitate to draw random shapes as well random is always fun while editing this video i'd like to add a couple of things with the graph in the shape of steps you can create a bit crusher effect we'll see bit crushes in a dedicated video but i thought it was important enough to mention it [Music] also a saturation reduces the dynamic of a sound as we basically turn the volume up until the signal hits a wall the quieter parts of the sound will be brought up and be closer in volume to the louder parts so we can use it to make discrete things in the sound more prominent or to squash several sounds together and make new textures appear like if we play several sine waves together at different pitches they'll go in and out of sync due to face cancellation it will make the sound appear and disappear well you can make that more obvious with the saturation this is the principle used to create wrist bases several sawtooth waves are detuned then filtered to keep only the low end and then distorted together the phase cancellation creates the movement that is particular to this wrist bases [Music] another common move is to play a sine wave with white noise and crush them together with a saturation to create a textured base [Music] there's one last question i would like to answer before i wrap up this video the last but not the least where can we find these wave shapers in ableton live serum and vcb rack in ableton live the saturator effect shows the curve used for each preset and with the last one called wave shaper you can access several parameters to design your own curve if you want more control to design your own curves i like the veggie wave shaper which is a max for live device that you can download for free it allows you to add as many points as you like use curves you have symmetric and asymmetric modes and you can lock the first and last points if you don't want them to move or you can snap the points you add to a grid for more precision it is very cool and certified without meat in serum in the effect tab the distortion effect also displays a graph which shows the curve used for each preset it's interesting to see that the drive knob on this plugin doesn't amplify the incoming signal but changes the shape of the curve instead [Music] in the preset list you have x-shaped presets in which you can draw your own curves directly by clicking on one of those two buttons here there is a symmetric mode in which the graph represents only the top part of the curve or asymmetric mode in which this represents the whole curve in this x-shape mode the drive knob still doesn't amplify the input signal but it is more of a transfer node between the shape of the curve a and the curve b [Music] then in vcv rack i haven't found a wave shaper that allows you to draw your own curves in this way instead there are wave shapers with various presets to choose from my two favorites among the ones that i've tried are the wave coast wave shaper by lindenburg research and shaper by squinky labs [Music] [Music] [Music] [Music] there is kind of a myth around compressors saying that it turns everything louder while it is kind of true in some situations it is not really how a compressor works in this video we'll see exactly how a compressor works what parameters you have to control it and different kind of compressor you can find a compressor is actually designed to make things sound quieter by turning down the volume of the loudest part of the sound so the dynamic of the sound which is the difference in volume between the loudest part and the quietest part will be reduced the dynamic would be compressed hence the name compressor and because squared part of the sound will be closer to the loud parts in terms of volume everything would sound closer to the ears and it can also bring forward some details in the sound that would otherwise be in the background so there are five parameters you can find on any compressor the threshold which is the point above which the sound will be compressed if the sound is not loud enough to reach the threshold then it won't be compressed then the ratio which sets how much the sound will be compressed the higher the ratio the more compressed the sound so for example if a sound exceeds the threshold by 10 decibels and the ratio is at 2 to one then this excess would be divided by two so the sound will only peak at five db above the threshold if the ratio was set to four to one the sun would be compressed by a factor of four so it would peak at 2.5 decibel above the threshold then is the makeup gain or output gain which allows you to make the overall sound louder to make up for the loss and volume due to the compression then there is the attack which will set how fast the compressor will kick in after the signal past the threshold so with an attack at 0 millisecond the sound would be compressed instantly and with an attack at 100 milliseconds the compressor would go from 0 to full compression in 100 milliseconds after the signal reached the threshold this is useful to let the transient of a sound go through before the compressor kicks in we'll see that in a moment and finally the release which tells you how fast the compressor falls back after the signal have returned below the threshold so with a long release the compressor will stay active for a little while it can be hard to actually hear the effect of a compressor at first so don't hesitate to crank up some knobs to get the other settings right before turning back down the first knobs you cranked up for example you can crank up the ratio and turn the threshold down to exaggerate the effect and hear it more easily then it can be easier to set the attack and the release to values that you like [Music] then turn the threshold back up to affect only the part of the sound you want to affect [Music] and then turn the ratio back down to a more subtle value [Music] and finally you can turn up the makeup to taste [Music] the thing is when you compress the sound it will fill the frequency spectrum more consistently because the quiet elements of the sound will be brought up relatively to the loud elements so the sound would appear thicker this can be used to make an element cut through a mix without making it actually louder but it can also make it clash with other instruments in the mix so there's a fine balance to find there now let's see the effect of a compressor on a kick drum let's be a bit extreme with it so we can hear better what it does so i'll put a high ratio and a low threshold now let's hear how the attack affects the kick drum as you can hear a very short attack actually makes the key quite dull it doesn't hit as hard that's because with a short attack the transient so the very first part of the sound that is really high in energy is brought down by the compressor so to keep the energy of the kick we need to let the transient through before the compressor kicks in so with a longer attack the transient have the time to go through before the rest of the sound is compressed and then you want to adjust the release it sets how long it takes to the sound to go back to normal volume once it went back under the threshold it all depends on the tempo and the feel of the track so it is easier to set this one in context with other instruments playing listen carefully trust your ears and stop when it feels good so the rule of thumb to remember is that generally a long attack with a short release is good to keep the dynamic of a sound so the transient have time to go through before the compressor kicks in which preserves the energy of the sound and the short attack with a long release is good to control the sound if you feel it needs to be more consistent it is often a matter of finding the right balance between the two so now this is the main principle of a compressor but there are different kinds of compressors or several ways to use them you could use it as a parallel effect which means you would split the signal in two tracks to have one channel that is processed by the compressor and one channel that is not one way to do that is to use a return track you put your compressor on the return track and then send your track to this written effect this way the sound of your track will be layered with the effect in your return track another way to do that in ableton live is to create an effect track with two chains one with no effect and one with a compressor this way you would have your original sound that is doubled by its compressed version which can be a good way to add beefiness to a sound like a kick drum for example [Music] you can apply a compressor to a group of tracks to affect several instruments together this can help tie up those tracks together in a more cohesive ensemble it can make it sound more like it's one block that is well held together you could use group compression to glue all your drum tracks together or all your lead tracks or to glue your base with the kick drum for example [Music] this is more on the mixing side of things but compressing a group of tracks can also highlight frequency issues so use it in combination with an eq on each track of the group so you can avoid having frequencies on one track clashing with the same frequency range of another track compressors designed to be used on a group of tracks are often called glue compressors but nothing prevents you from using a glue compressor on a single track or using a regular compressor on a group they are all essentially the same thing you could also split the signal into different bandwidth to have only the lows in one channel only the mid in the second channel and only the trebles in the third one this way you could apply different compressions to each channel some compressors have this split channel system already built in they are called multiband compressors they work like several compressors one by frequency bands and usually they let you set the split point between two bands where you want so they are very handy to have more control over your compression if you need that level of precision ott is a particular preset that got so popular i thought i needed to include it here ott stands for over the top and it was originally a preset for ableton's multiband compressor but then it got so popular that it's been made at the free third party vst plugin by exfer the guys who made serum and it's really weird because it really squashes the audio like there's no tomorrow which would be a bad idea in a lot of contexts if you do that on every track it will feel like you brought every sound closer to your ears and every sound would be competing against each other that would end up in a very messy mix but in the face of sound design you can use it like a magnifier to reveal all the textures of the sound that is normally in the background because it compresses the lows the mids and the trebles independently and it compressed them so much everything will be brought up to the same level and in some music genres it's become a common practice to put several in a row if you really really want to squash that sound so in the end it's a very nice thing to try out and then you can filter out some frequencies afterward because it will really push up all the frequencies in the spectrum [Music] i think it works better on same sounds than recorded samples because whether recording it will really reveal and bring up all the background noises [Music] on the ott the main parameters would be the time knob which acts a bit like a combination of attack and release [Music] and the amount knob which acts a bit like between a general ratio and a dry wet the amount knob is not really a dry wet though if you put several otts in a row with an amount of zero you will still hear a transformation in the sound i think it is due to the way the multiband compressor separates each banner frequency but anyway that's still a cool thing to try and then you have the level of each band that you can use as a small eq the ott is a particular case it's not for all sounds but is very popular so we'll probably talk about it again another way to use a compressor that is very popular in electronic music is to use it in sidechain when a compressor is set to sidechain it takes an external signal as an input and then it is this external signal that will trigger the compressor the most common application is to put a compressor on the base and put the kick track in the sidechain input so every time the kick plays it will trigger the compressor which will compress the base so the base will duck every time the kick plays creating a pumping effect personally i usually prefer to use an extra track dedicated to trigger those sidechain compressors basically i would create a midi track and create a synth that makes a very clicky sound by turning down its envelope's attack sustain and release and then setting a very short decay it's very easy the kick is already played by a midi instrument so you just have to duplicate this midi track and then i mute this new midi track and use it to trigger my search and compressor because the sound of this synth is so short it's easier to make the release synced to the tempo you would put the attack very low so the compressor engages right away when it received the click and then the release sets how long it takes to the base to come back at normal level so if you want that time to be of a quarter note just take 60 000 which is the number of milliseconds in a minute and divide it by your tempo which is the number of quarter note in a minute it will give you the time for one quarter note which you can put as the release time this way you can have a sidechain compression synced to your tempo [Music] another way to mimic that in ableton for a four to the floor rhythm that i find cleaner sometime is to put an auto pan effect it's a panning effect but if you put the phase at zero sync it to the tempo and use the reverse sawtooth waveform you can mimic the pumping effect then you can play with the shape knob to change the way the bass is coming back up here i'm showing it whether based on a kick but it can help in a mix for any instrument to set priorities for example you can duck some tracks down when the singer's voice is coming in a compressor can also be used as a de-esser which can reduce the sibilances of a voice recording it's used to reduce the level of the sur and sure sounds to do that you need some kind of multiband compressor to compress only the frequencies where the sibilances live usually the sure sounds are around five thousand to six thousand hertz and the s sounds are usually around seven thousand to nine thousand hertz seven thousand to nine thousand hertz seven thousand to 9000 hertz in ableton's regular compressor when you open the sidechain with this little arrow you also have access to an eq section there you can select a bandpass mode so only a bend or frequency will trigger the compressor this is how the dsr preset has been built but if only the sibilances will trigger the compressor it's the whole sound that will be compressed but it is still a good way to lower some of the harsh parts of the sound a limiter is usually identified as a separate effect but it works in a very similar way than a compressor if you crank the ratio of a compressor all the way up until it says infinite ratio you have what is called a brick wall compressor this way no sound would go past the threshold because any sound that would go past it would be brought down at the threshold level and this is exactly what a limiter does it prevents anything from growing higher than a certain level usually it is used on a truck on the master bus with the threshold at 0 db to make sure you don't get any clipping overall a compressor will reduce the dynamic of a sound while bringing it closer to the ears of the listener maybe this video was more on the mix inside than actual sound design but i hope it's been useful to some of you compressor is kind of a mysterious effect you hear a lot about it but it's kind of hard to actually hear what it does or at least that's something i struggled a lot with when i was starting out reverberation is the result of sound waves bouncing off many surfaces such as the walls the floor or the ceiling it is actually present in any sun we hear in our everyday life so we don't always pay attention to it but when a sound is played in a room or in any space really we perceive the direct sound from the source followed by the reflection of that sound of the surfaces in that space when the time between these reflections is long enough it creates an echo effect where we can hear the original sound repeating several times this is what the delay does which is an effect we'll see in the next video but when the time between these reflections is short and scattered enough they blend and create kind of a trail of sound this is what we call reverb so reverb effects are audio effects that are designed to emulate this type of reflections and it is good to place your sound in a particular space in a studio environment when we record an instrument or voice we often try to eliminate as much of these reflections as we can with acoustic treatment like foam panel for example but in the context of a song a dry recording without any reverb can sound thin or natural so why do we tend to eliminate all this natural reverb when we are recording in the studio well it's because it allows you to clean your recording before it goes to a reverb because a reverb would take all the frequencies in the sound and kind of scatter them in the frequency spectrum so if you can remove unwanted frequencies or noises before applying any reverb effect it would be easier to have a clean mix so reverb effects are designed to emulate the reflections of a sound in space and there are several ways to achieve that chamber reverbs were originally made by placing a speaker and a microphone in a room that's where you would directly record the sound reverberating in a real room the name of the effect would then change depending on the size of the room studio reverb for a very short one a room river monitors a little longer up to a whole reverb for a very big space nowadays it is way easier to recreate this effect with softwares which we'll get into in a minute or you could create a plate reverb by playing a sound through a metallic sheet and then recording the resulting sound with contact microphones attached to the plate basically imagine a speaker that instead of vibrating a cone to produce waves of different air pressures it makes a big sheet of metal vibrate so the metal sheet will distort altering the sound in the process and that creates an effect very similar to the reverb of a room but there are still some differences between a plate reverb and a chamber reverb though sound travels way faster in metal than in air particularly for high frequencies so with the plate reverb which makes the sound travel in metal you will hear the higher frequencies slightly before the lower ones and because of optical acoustic phenomenon called precedence we hear the overall sound as brighter so plate reverbs are cool to make something sound brighter without actually boosting the high frequencies and that's why it's used a lot on vocals or acoustic guitars for example and the fact that sound travels in metal faster also makes the time between each echo way shorter which make the reverb more dense and also more consistent over time in a similar way you could run the sound through a metallic coil to create a spring reverb which is a popular effect for guitars when the sound travels through a spring it bounces back and forth which creates a succession of rapid echoes and it's the succession of these echoes that makes the tale of the sound that make it sound like a reverb the fact that the sound needs to travel the distance of the spring back and forth to create each echo makes the time between each echo longer which give the spring reverb kind of a bouncy quality the way the spring vibrates also affect the frequency response of the signal which gives the spring reverb its particular sound in fact sound waves need to travel along the coils of the spring so they move the chords one after the other to move forward and low frequencies with the longer wavelength can make several coils move at the same time which allows lower frequencies to travel faster than higher ones which means we'll hear the lower frequency before the high ones so the resulting sound will seem darker now not everybody have the space or equipment to create this kind of reverbs mechanically and this is where softwares come into play there are actually two ways of creating a reverb with softwares there are algorithmic reverbs and convolutions reverbs algorithmic reverbs are software that works only from their own internal algorithms so they are generally less accurate when it comes to reality them but they are also lighter on the cpu they can be designed to replicate a chamber reverb a plate river by spring reverb but they can also be designed to do more creative stuff like a shimmer reverb that pitches up every echo slightly so the trail of the reverb would go up in pitch giving a brighter tone [Music] all the other way around the trail could go down in pitch to give a darker tone [Music] [Applause] [Music] now convolution reverbs on the other hand work in pair with a particular type of file called impulse response which is like the fingerprint of a room or a piece of equipment that exists in the real world [Music] basically to create an impulse response of a room for example you take a very short impulse sound like one sample long a very short burst of white noise then you play it in the room you want to create an impulse response file for and you record the reverb with the microphone this recording you made is the impulse response then all you have to do is to put it in a convolution reverb and it will emulate quite accurately the reverb of the room you recorded in convolution reverbs are actually the most accurate when it comes to recreating reverb from a real world place and the beauty of it is that you can also generate impulse responses for other equipments than a room you could create an ir file for spring reverb for a plate reverb or even for guitar amplifiers softwares that emulate guitar cabs use the exact same technology and it goes even deeper because impulse responses are typically wav files so you could load your own sounds in a convolutional reverb to see how it reacts and use it as a sound design tool [Music] [Music] this is a deep rapid hole to explore so i may do that in a separate video as this one will be quite long already so in ableton you can find a free convolution reverb as a max for live device simply called convolution reverb it's part of the essential pack that comes with it and it already has a lot of impulse responses in it but you can still add your own if you want to have more or if you want to use it as a guitar cab emulator so how do we actually use these reverbs we are going to see the main parameters you're likely to see on every reverb and how they affect the sound and then we'll see some tips and tricks to get the most out of them i'll be using ableton stock reverb for the demonstration which is an algorithm reverb so there you have a dry wet knob that allows you to mix the dry signal not affected by the reverb with the wet signal and then the main parameter of any reverb i guess would be the decay time which will set the length of the reverberation [Music] next to that you also have a stereo knob to make the reverb sound more or less white in the stereo field [Music] and a size knob to emulate either a small room or a bigger room [Music] so generally a smaller size would work fine with a smaller decay time and a bigger size would work fine with a bigger decay time but you could also make more surreal effects with a small size with a big decay time for example note that this reverb also have a setting of economic medium and high quality that can change the way the reverb sound so don't hesitate to play with that as even on high quality it is quite cpu friendly another parameter you're likely to see on any reverb effect is the pre-delay knob it is basically a delay time between the dry signal and the first echo of the reverb a short pre-delay will make the sound appear closer to the walls of the simulated room and a longer pre-delay will make the source of the sound appear farther away from those walls and closer to the listener [Music] and lastly i want to talk about the reflect and diffusion knobs the reflect knob often called early reflection sets the level of the first group of echoes these early reflections are usually more defined than the rest of the tail it sounds kind of more like a delay than an actual reverb so turning up the early reflection knob usually works better on sustained sounds like vocals or pads and turning it down places the sun further back in the room which often sounds better for more percussive sounds [Music] and the diffuse knob sets the level of the tail so you can have more control over the balance between the tail and the early reflections so i won't go in details in all the features this reverb has because it is not a tutorial on this particular reverb but instead let's see some tips on how you can use reverbs in general so how to use reverb one thing to keep in mind is that every frequency will be scattered in the frequency spectrum once it's processed by a reverb so it is a good idea to place an eq before the reverb to get rid of problematic frequencies it's always easier to clean a sound before the reverb than after and you can also put an eq after the reaver to make it sound as you like that's why in ableton's reverb you have two simple eqs one to filter the sound b for the effect and one after reverb is a nice effect to make a sound wider by putting the sound in a space but when you use reverb on a sound it also makes it appear more distance farther away from you from the listener so it is a balance you want to keep in mind while using reverbs one thing i like to do to have more control over the reverb is to create a parallel track for it so basically in ableton i would create an effect track with two chains one chain with no effect that would be my dry channel and one chain with a reverb set on 100 wet then i would play with the volume of each chain to set the balance i want [Music] you can also link one of the macro knob to the chain selector to create a new dry wet knob for this effect track [Music] having these two separate chains allows you to process the dry sound and the wet sound separately and more freely for instance you can put a compressor on the wet channel to make the reverb a bit thicker [Music] or you can even use sidechain compression and take the drive signal to trigger the compressor this way the reverb will appear only after the dry signal have stopped playing because the dry signal will dug down the river when it plays and then you can play with the release of this compressor to adjust how fast the reverb comes back in after the dry sun has stopped playing [Music] you can also add your own eqs before and after the reverb which will give you more control than a built-in eq and it is really useful to control the color of your reverb so it would fit the mix better for example if you have some prominent frequencies in the original sound that can cause problems once it's put through a reverb you can tame them beforehand and then the second eq is there to shape the overall color of the reverb speaking about eq i often prefer to cut the lows in the reverb as reverb and low frequencies tend to cause chaos in a mix but there are some exceptions to that you can also use a reverb on a return track so you could send any of the tracks of your song to that reverb and each one with a different value so you would have all your tracks unaffected by the reverb layered with the reverb version of themselves and because these reverb versions all come from the same reverb it's a good way to place all your instruments in the same space and this common space these common acoustic characteristics would tie everything together in a coherent way [Music] [Applause] [Music] [Music] and because this written track is also its own chain you can also use eqs compressors and other effects to shape this reverb as you like these all work in the context of both mixing and sound design but on the sound design side you can also simply use a reverb to add a tail to a very short sound like a very short drum sound to make it a bit longer some reverbs also have a freeze option it will hold the sound of the reverb so you can have a sound that sustains forever it's cool to have a long sound to resample and to use as a starting point for a new sound design [Applause] another thing i like to do that is less conventional and that i took from techno music is to use parallel reverb on a kick basically you would create two parallel chains with an effect track just like we did before with a dry channel and a reverb on the wet channel and then you want to add the low pass filter before the reverb so it will affect only the low frequencies this is a good trick to add some boom and some umph to a kick and i found it sometimes more effective than parallel saturation or parallel compression to give this extra thickness i wanted this is one of those weird tricks as generally putting a reverb on low frequencies is considered a bad practice because you want to have the bottom line of your mix in mono but you can still add a utility effect and drag the width knob to 0 to make it mono again and with this one it can be good to use the sidechain compression triggered by the drive signal this way you would hear the rumble after the kick has stopped playing to keep it a bit cleaner the thing to keep in mind here is that it would add low frequencies so if you have a bass playing in the same range the two could clash so you could keep this trick for a track where there's not too much happening in the very low end or you could have an eq that cuts the low end of your base so it doesn't clash too much with it one last thing i wanted to add is a very popular trick on how to use the pre-delay i think it's too important not to include it here let's take this chain we've made for the kick as an example we could use the pre-delay to make the reverb come a bit lighter to add this bouncy feel to it and sometimes you want to have this pre-delay time synced to your tempo so it would kick in after exactly a 16th note or edge note and if we want to do that we need to find how many milliseconds exactly is that 16th or 8th note here's how to calculate it easily your bpm gives you the number of beats that are in one minute these are the quarter notes you also know that in a minute there are 60 seconds that's 60 thousand milliseconds per minute so we need to divide this sixty thousand milliseconds by your bpm to have the length of one quarter note at 120 bpm the quarter note is 500 milliseconds divide that by two and you have the length of the eighth note that 250 milliseconds and divide that by two again and you have the length of a 16th note that's 125 milliseconds now let's hear how the reverb sounds with a pre-delay of a 16th note so i'll put 125 milliseconds here now with a predelay of the eighth note so that 250 milliseconds so you can use this pretty lineup to add some bouncy nets to your tracks using it in this way essentially makes a slap back delay effect which is something we'll see very soon in the next video when we talk about delay effects when a sound is played in a room or in any space we perceive the direct sound from the source followed by the reflection of that sound that bounces off the surfaces in that space when the time between these reflections is long enough it creates an echo effect where we can hear the original sound repeating several times [Music] this is what a delay effect is designed to do a delay effect will repeat any sound it receives it usually have two main parameters a delay time and a feedback the delay time is the time between each echo and the feedback is pretty much the number of those echoes for example on ableton's simple delay you can set a different delay time for both channel left and right they can be by division of the tempo or in milliseconds or you can link the channels left and right to use only one delay time then like on any reverb you would have a dry wet knob to mix the original sound with the delayed sound when playing with these parameters you can create different textures or different sounds create different effects for different uses so that's what i'd like to show you now the most simple use of the delay would be to use it as a rhythmic element by repeating any pattern it receives you can then create a more complex pattern starting with a very simple one delays are also great to add space around your sound so it is a very good companion to a reverb effect they are often used hand in hand as they both simulate reflections we can find in raw spaces so with a subtle touch of it you can make something sound a lot bigger [Music] if the echoes happen once on the left side and once on the right side alternatively you end up with a ping pong delay you can mimic this effect by setting different delay times for the left and right channels but most of the time you would have a ping-pong mode in your delay here in ableton it comes as a different effect if you set the feedback very low you should get only one echo and then if you have a quite short delay time you would have what is called a slap-back delay it is used a lot in the context of a mix on vocals or guitars for example it adds a good sense of space without washing out too much of the dry signal [Music] if you make this delay even shorter the dry signal and the echo will start to blend and you can then achieve kind of a doubling effect that can make the sound thicker but be careful with the phase cancellation when you do this some frequencies may become silent when the dry signal and the echo mix so you may want to fine tune this [Music] you can also put the dry signal on the left and the echo on the right or the other way around to create a white stereo effect or on this delay you can put the dry wet all the way up so we hear only the echo and the feedback all the way down so there's only one echo then set one side to zero milliseconds so it's basically the dry signal and the other side a few milliseconds later this will make your sound a lot wider [Music] now let's go back to mono but let's keep a very short delay time if we increase the feedback we increase the number of echoes and as they are very close to one another they will create friction this will make the original sound resonate on a certain frequency and that can make it sound like some robot voice [Music] do [Music] [Music] it can be very cool to make the note resonate so you can resample it but you will most often have to retune the notes as it's very difficult to tune it with the delay effect directly but if i'm not mistaken that is the same technique that is used by the effect resonator in ableton and in that one you can choose the note at which it will resonate and you can even layer several of them to make a chord [Music] that's already several ways you can use any delay effects to create different sounds but some delays are made more complex than others some delays add one or several effects before or after repeating the echoes such as the tape delay the tape delay is a delay effect that makes echoes but before repeating these echoes it distorts the sound with a very smooth very soft saturation to add warmth to the sound and it was originally made with analog tape machines so here we'll try to replicate several aspects of this tape first let's add a delay and add a tape saturation to mimic the tape's warmth the old tape machines didn't have a sync button so it was near impossible to have something really locked to the tempo and that gave a particular swing to it to emulate that we can either use a rate time in milliseconds to have it not synced or we can have a synced rate but offset the whole thing by a few milliseconds [Music] the old tape machines were kind of low fidelity they used to compress the sound a lot and be quite poor in the high frequencies so for the compression part we can use a compressor with a high ratio and a low threshold with a quick attack and a long release to mainly take down all the peaks and then we can also add a low pass filter to cut some of the high frequencies this will have a double advantage of letting some room for the unfiltered sound layered with this effect and pushing the delay motor at the back of the mix to give it a good sense of space all tape machines also used to have an irregular speed of playback which would introduce shallow pitch fluctuations in the echoes and that's something we can also emulate with a delay effect you can right click it and select repeat this way if we move the delay time it will pitch the sound up or down which is basically the behavior of the tape that goes up and down in speed so instead of moving this by hand or automating it i will control it with an lfo that goes slowly up and down automatically it doesn't need to go up and down very far our small values are enough for the saddle effect we want [Music] now we have a chain of effect designed to replicate the grit of the tape its compression frequency response with the pitch fluctuation and the time irregularities now this kind of effect often sounds better as a parallel effect by layering a sound processed by this effect with the same unprocessed sound so to do that let's group all that then we can either put it on a return track say return track a and send any track to return a or you can simply create another chain in the group with no effect in it so any sound that goes in this rack will go in both chains you can then adjust the balance between the two with their volume there [Music] [Music] so in the same way you can create your own composite delays by combining any effect you want saturation flanger etc and you can even create several parallel chains to layer different delays for example let's make a delay that uses several other effects let's make a dub delay dub delays usually combine a very long feedback with a lot of effects for a very lush result usually with a lot of movement in it so to do that let's use a return track so let's put a ping pong delay for once on the return track a with a long feedback and a dry wet at 100 as it will be layered with a dry sound anyways and then we send an instrument into it the cool thing about return tracks in ableton live is that you can send the return track back into itself to create a feedback loop that's what we're going to do but before let's put a limiter at the end of the chain to protect our speakers because feedback loops can go out of hand pretty quickly and then we can send the return track a back into itself [Music] to add motion we can use a filter with a lfo we could use the built-in filter in the ping pong delay but for this one let's use the auto filter so we can use the built-in lf it has or for a more complex sound we could create another delay on another return track so the return track a will be sent into return track b which will be sent back into written track a but don't forget to add a limiter at the end of each chain though [Music] then to add movement you can automate any parameter of any effect you would add in including the knob sending the sound to return a and return b or you can also put the delay in repeat and play with the delay time [Music] but now i would like to have a closer look at the last thing we did with our delay effect we loaded the delay effect we right clicked it and we chose the repeat mode you can use this mode to automate the delay time to create either an ascending or descending pitch and that's a great technique to create risers ambient textures [Music] [Applause] [Music] there is a pretty similar effect that is used a lot in dub music for example it is not really a delay effect but as it's pretty similar let's include it here you can use a sampler as a looper basically you take a sample most of the time it's a sample of a voice you put that in the sampler and you put it in loop mode this way when you play a long note the sample will play in loop it is not really a delay effect it is not really a delay effect and then you can automate the pitch of that sample of the pitch of the note it is not really a delay effect it is not really a delay effect it is not really a delay effect it is not really an effect this way the sample will play higher and higher but it will also play faster and faster and it's really effective to make a good riser it is not really a delay effect it is not really a delay effect it is not really a delay effect it is not really the effect as we just saw delays are great tool to find new textures or to add space to a sound and delays are rarely used on their own to give a sense of space they are often used with a reverb and like reverbs they are often used as parallel effects it is a great effect with both timbral and rhythmic qualities in the last video we talked about the delay effect and a lot of different things we can do with it in today's episode we'll follow the same direction as we will explore three effects that use a very similar process we'll talk about flanger chorus and phaser effects they are called modulation effects because they usually have one or several lfos inducing movement in the sound [Music] so let's begin with the flanger so this is a sawtooth waveform alone now with a flanger but without the lfo and now with the lfo when a signal goes through a flanger it is duplicated into two copies one is played back as is and the second is delayed slightly it's generally a very small delay time like below 15 milliseconds late compared to the dry signal so up to now it's really like a delay effect these two signals are then layered and this creates some phase cancellation and amplification at some points for some frequencies the two waveform will both have a positive value of both negative values so their values will be added and make the signal more powerful and at other points for other frequencies one will be positive while the other will be negative so they cancel each other out as a result some frequencies in the sound will be amplified and other will be cancelled to attenuate it we can then trace a graph to show which frequencies are boosted or which one are cancelled and if we do so we get something like this this is the same shape that a comb filter have and that's why some comb filters have the name flanger attached to them and because the second signal is uniformly delayed these frequency cuts are uniformly spread out along the frequency spectrum it follows a harmonic series then if we move the delay time of the second signal we move this harmonic series so it's just like if we move the cutoff of a comb filter [Music] on many flanger effects you should also have a feedback parameter that sends the result of the effect back into itself and that basically accentuate the effect of the filter and on some flangers you also have an option to choose the polarity to create a feedback or fit4.com filter you can check the episode about filters for more info on that [Music] now the flanger effect is often associated with a sweeping sound and that's because usually the delay time is controlled by an lfo that makes it move up and down so the lfo will move the equivalent of this filter up and down creating this whipping effect [Music] [Music] so to sum it up on a flanger effect you often have a delay time and a feedback parameter to basically set the position and the strength of the comb filter and then you will have a rate and a depth knob which will define the speed of the lfo and how far it will move the delay time now let's hear a chorus effect [Music] a chorus effect is very similar to a flanger effect when a signal goes through a chorus effect it is also duplicated the first one is also played back as is and the second one is also delayed but generally it's delayed by a higher amount between 20 and 50 milliseconds then a chorus effect also generally has an lfo that modulates the delay time of the second signal just like in a flanger but unlike a flanger modulating the delay time will also alter the pitch of the signal just like with the delay effect when we right click it and put it in repeat mode shortening the delay time will make the pitch higher and making the delay time longer will make the pitch lower so with the lfo the pitch of the second signal will go up and down just like a vibrato so mixed with the dry signal it makes a dissonance and as any dissonance that creates kind of a beating in the sound if the effect is very drastic it can make a straight up probably dissonant sound but with more subtle values it is often perceived more as a rich hammering tone [Music] in terms of frequency response when we mix the delayed signal with the dry one it also makes the effect of some kind of comb filter but different than a flanger it is also possible to have more than one copies of the original sound to have several delays with different settings to accentuate the effect because usually there is no feedback control on a chorus i mean some of them do have feedback control but it's not always there uh [Music] it's an interesting effect that gives the impression that the sound is played by more than one source hence the name chorus and it is often used on vocals pianos or guitars to add thickness or interest to the sound so to sum it up on a chorus effect you would have a delay time parameter just like on a flanger and then you'd have a rate and an amount knob to control the speed of the lfo and how much it moves the pitch of the second voice also some chorus effects are called stereo chorus this stereo effect is achieved by delaying the left and the right channel differently with two different lfos that have different speeds so it will pitch if the left and the right channel differently as well now let's see the phaser let's see how it sounds [Music] a phaser effect is also very similar to a flanger effect but it's a bit more complex the signal that enters the phaser is also duplicated with one signal being played back as is and the other being processed the processed signal is kind of delayed as well but it's delayed differently than in a flanger effect the signal goes through one or several filters that are called all pass filters an all pass filter doesn't really cut any frequency what it does is phase shifting the signal in a non-linear way basically that means that it delays some frequencies in the sound differently than others so you can have the base frequencies delayed more than the higher frequencies for example so when you mix this signal with the dry signal you still get some face cancelling you still get some kind of a comb filter effect but compared to a flanger effect a phaser would have less cuts sometimes even just one the frequency cuts do not necessarily follow a harmonic series as a result a phasor effect will sound less harsh than a flanger effect because the effect is not tuned to a particular frequency so it will sound less drastic [Music] actually the less all-pass filters in your phaser the softer the effect and the more all-past filters used in your phaser the more frequency cut you add some phasers even let you activate or deactivate these filters they often refer to it as poles or stages so on a phasor effect you generally have a frequency knob and a feedback knob to move the series of cuts and their strength and you'd also have an lfo with a rate and a depth parameter and that would modulate the frequency so that can create a sweeping effect close to a flanger but less harsh then you may have a pole or a stage parameter to use more or less filters in your phaser so the thing to remember about the flanger the chorus and the phaser is that they all use a short delay time to alter the sound and they all have an lfo to give it some movement and it is this lfo that gives them their status of modulation effect personally in a sound design phase i like to use them as static effects without using the lfo so i would turn its amount to zero [Music] mom [Music] i like the texture it gives but on the opposite you can use a very fast lfo past a certain rate you can't really hear the oscillation of the lfo when it becomes some kind of a distortion just like in fm synthesis where you oscillate the pitch of an oscillator so fast that it just becomes a distortion yes [Music] foreign or you can simply use a normal or slow rate for the lfo to simply have motion in your sound another thing to keep in mind is that the result of these effects are the equivalent of adding a comp filter which means subtractive synthesis that means that this effect would work better on sounds with a rich harmonic content or at least you will hear the effect more on sound that have a lot of harmonics so it can work well on sounds that are generated with fm synthesis or that has distortion on them for example a bit crusher is a form of distortion and its purpose is to lower the quality of a digital audio signal it creates some recognizable lo-fi effect reminiscent of all gaming consoles like the game boy or the super nintendo but it can also be used more subtly to add a bit of grit or a bit of warmth to a sound just like any distortion effect really any audio signal when it's recorded into a digital format will be cut into small chunks each chunk will hold the value and when you put all these values together you get the waveform of the sound you recorded it's exactly like a photo you see on your computer screen it's made of pixels and each pixel holds only one color let's take a moment to see how that works a bit more in detail it will make everything easier to understand the bit crusher when the sound is cut into slices the number of slice per second is called the sample rate and the default sample rate in the majority of music software is 44 and 100 hertz so that's 44 000 and 100 slices per second that's a big number and that's because to record a note properly we need a sample rate that is at least twice as fast as the frequency of the note we want to record that is called the ninquest limit so we can have at least the two peaks of the waveform and we can know its frequency on average the highest frequency a human ear can hear is around 20 000 hertz so we need at least a sample rate of 40 000 hertz to record it so with the sample rate of 44 and 100 hertz we have a little room to be safe i open a quick parenthesis other little trivia in video the standard sample rate for sound is 48 000 hertz and that's because in cinema we have 24 frames per second so having a sample rate that is a multiple of 24 made it easier to synchronize the sound with the image end of the parenthesis now for each slice it will record a value and the precision of that value is called the bit depth like in any computer this value will be noted as a series of zeros and ones don't worry you will never see those numbers it's just the software doing its thing but the thing is the software will have a limited amount of slots to write the value for each slice and these slots are called bits if it reads the value on 2 bits it will only have 4 different options to write it 0 0 0 1 1 0 and 1 1. so the more bits you have to write these values the more precise it will be and generally for a wave file the bit depth is of 16 or 24 bits which is a definition of respectively 65 000 and 16 million different possible values for one slice now the purpose of a bit crusher is to lower the quality of an audio signal and this is exactly how it will achieve that it will reduce the sample rate and the bid depth these are the two main parameters you will see on almost all bit crushers so what does it mean for the sound when you lower the bit depth it will introduce some stepping in the shape of the waveform and changing the waveform means changing the harmonic content so this will add harmonics above the fundamental the more you lower the bit depth the more the waveform will look like a square wave and that's what will make your sound sound more like an old video game [Music] for example the sega mega drive or genesis had sound in 8 bits and other up to 12 bits and game boy had them in 4 bits but don't let this fool you you can make very powerful sounds at a little bit depth the ronan tr-909 for example which is an iconic drum machine had a bit depth of only 6 bits lowering the bit depth can also reduce the dynamic range because you'd have less possible values between the minimum and the maximum amplitude so you can lose some subtle differences in the volume and it can also bring up the volume of the noise floor compared to the volume of the signal and it can also introduce some quantization noise which sounds like a white noise with a low pass filter on it [Music] if you want to have a cleaner sound it's sometimes better to boost the volume of the sound that goes into the bit crusher so the dynamic would be higher [Music] because of this noise that can be brought up i prefer to have this kind of effects on synthesized sounds instead of recordings and when you reduce the sample rate which is also called down sampling you will introduce a lot of artifacts to the sound as said before to record a frequency properly you need a sample rate of at least twice that frequency so when we reduce the sample rate some higher frequencies in the sound will become simply too high for the sample rate and because we don't get enough samples to describe them they will be misinterpreted as lower frequencies and this is what we call aliasing so it will add in harmonics to the sound in completely random places even below the fundamental this creates a particular sounding distortion that is very characteristic of digital [Music] [Applause] and by the way a lot of digital distortions suffer from the same aliasing problem because the distortion at high harmonics some of them will be too high for the sample rate and so they bleed back as lower frequencies so if you want to avoid this aliasing sound you can use a low pass filter just before the bit crusher filtering out the higher frequencies will prevent them from causing problems so you can get a much smoother distortion this way you can even make it sound like an old tape recording [Laughter] [Music] e so a filter is a really good companion to a bit crusher this trick is cool to create some more subtle effects that can help a drum track or a vocal tract to pop out a bit more for example [Music] you can even try to put an lfo to control a low pass filter or a bandpass filter so it will add and remove some random harmonics from the aliasing and it will create a distinctive oyo sound ai if the original sound is a static waveform lowering the bit depth will create a waveform that is static but with steppings whereas reducing the sample rate will create some stepping in the sound but with motion and because the speed of this motion depends on the frequency of the note and the sample rate it can be a good idea to resample it if you want the same motion for every note hi i am ucha and this is sound design theory today i would like to talk about three effects designed to be used on voices and that are easily mistaken talking about them would be a good opportunity to talk a bit about formants as they all utilize formants to alter the sound the three effects we'll tackle today are the vocoder the autotune and the toolbox effects so we already talked about formants in the episode about filters but really what are they when a sound is played in a cavity that can be a room a box or whatever the object containing the sound starts to resonate and this will make some harmonics in the sound louder due to the frequencies of resonance of the object that it is played in so this will create peaks in the audio spectrum and it is those peaks that are called the formants so when we are talking it is exactly what we use to articulate vowels when we talk our vocal chords vibrate and then we use the position of our mouth and tongue to produce different formants that's what makes a e r and o sound different they can all have the same fundamental note but the harmonics are different and if the harmonics are different then the sound is different you can actually note the frequencies of this formants for each vowels and then you can use filters tuned to these frequencies to make vowels out of any sound that's exactly what we did in the follow-up video of the episode about filters i'll put the link there and in the description now that we know what formats are we can see more easily what a vocoder and an autotune effect are which are often mistaken for one another and then we'll see the talkbox effect the vocoder as an instrument looks like a synth with a microphone but it is now available as a vst and softwares so for this video we'll take ableton's vocoder stock effect as an example the purpose of a vocoder effect is to mix the sound of a voice with the sound of an instrument of on a synth so we would still hear that instrument but it would articulate words just like the voice [Music] so to do that we need two signals a carrier and a modulator one will be the base sound that we're going to hear so that's the carrier here it will be a synth and the second signal will change properties of the first one that will be the modulator and here it will be a voice [Music] so in ableton you would patch the vocoder effect on the track of the modulator so that would be the track with the voice and in the carrier section of the effect you can go get the track with the synth the way it works is that the vocoder splits both signals in several bands and then it will analyze the level of each band for the modulator and applies the same levels to the carrier sound you can see this as an eq effect that is applied to the carrier so its harmonic contents would follow the harmonic content of the modulator and that means a couple of things this will work better if the sound of the carrier is rich in harmonics because we're going to curve frequencies out of it this can be a synth with a sawtooth waveform a sound with a heavy distortion on it this kind of things this also means that the carrier will inherit the arrhythmic qualities of the modulator as well if the modulator is not playing no frequency band will be open so we won't hear the carrier neither so you could kind of beatboxing the mic to chop up a synth for example [Music] so as we talked about formants if the voice says a the vocoder will pick up the formants of that a and apply it to the carrier so the instrument will say e as well then some recorders will have a format knob that is pretty cool it will shift the format up and down so the tone of the voice will sound higher or lower but the pitch will actually stay the same it can sound more like some kind of filter and on a voice that's how you can change the gender for example [Music] [Music] now you can emphasize the effect of that format knob by playing with the settings of those frequency bands here you can set the range of frequencies that will be scanned by the vocoder right now it goes from 80hz to 10khz but if we make that trend narrower we should hear the effects of the format not better [Applause] [Music] you can also choose the number of bands that you want to use which can drastically change the sound and you can even set the level of each band if you want to use it as an eq and to clean the signal even further you also have a gate here so the sound would be processed only if it's louder than a certain level [Music] on this vocoder you also have a setting for the width of each bend which is pretty interesting at 100 each band starts where the next one begins but if you make that narrower you can hear how the sound gets a bit more fluttery as there will be less frequencies in each band it will make the sound closer to a sine wave or you can also make this band with higher than 100 in this case each band will overlap with the next one [Music] so this is mostly how the vowels will be processed because this will pick up mostly formants and formanced or mostly vowels so how do we get the rest of the signal how do we get the consonants well for that you should have a knob or a section called unvoiced with a knob that will allow you to bring back up those consonants it's the part of the signal that is pitchless like the sibilances that will be applied to a layer of white noise here the big knob is simply the level of that layer and the sensibility is a bit like the depth knob there it's the strength of the filter applied from the modulator to the carrier [Music] this is of course good to make a vocoder voice more intelligible but it's also fun to use on more rhythmic elements like beatboxing for example [Music] now if you want to experiment more on that kind of sound you can also select noise as the modulator in this case you won't need a separate track as noise will be used instead of a synth and this noise is generated directly by the effect you can then play with this xy pad to change the character of the noise on the x-axis it's some down sampling i explained what it is in the last video about this crushers and on the y-axis you can change the density of the noise [Music] you can try this mode on anything but i find it particularly cool on drum sounds especially if you play with the envelope and if you keep only the high end with the band eq you can also use it to enhance the transients of the drums [Music] do [Music] [Music] in the carrier mode you also have two other options here you can use the modulator as its own carrier so that will make a re-synthesized version of the same sound and that's mostly if you want to play with all the settings without actually blending the modulator with another sound which can also be great for sound design [Music] and you have a pitch tracker mode which will use an internal oscillator that will attempt to follow the pitch of the modulator so here the pitch will be given by the modulator instead of the carrier and you can use the high and low parameters to tell it where to search for the pitch to track [Music] then instead of changing the carrier you can also change the modulator it doesn't need to be a voice so if you have a synth that has movements in its pitch you can use it as the modulator to apply those movements to another synth or to a noise [Music] [Music] oh [Music] or you could use a very low sine wave as the modulator to mix it with the texture of a rich carrier to get a gritty base because remember the vocoder works with bends so if the sine wave triggers one bend it's the whole band that will be activated on the carrier and then you can turn down the depth knob so an even larger bend of the carrier can be hurt now the autotune is often mistaken for a vocoder as it can also produce a robotic voice but the way these effects work are totally different the autotune effect unlike the vocoder doesn't use a carrier and a modulator signals it uses only the signal from the voice it is affecting also even if it can affect formants it doesn't use formats in its core processing so how does it work the auto-tune effect is a pitch correction tool you run a voice through it it will analyze the frequencies that compose it and then it will give you options to correct the pitch of this voice so it matches the notes of the scale you want to use auto-tune is actually the name of a particular effect from the company antares audio technologies but this technology is now available in other competitor softwares like melodyne very audio wavetune isotope nectar's pitch correction vivocore etc actually i researched how exactly it processes the sun under the hood but it's still very opaque so let's see how to use it instead basically you put auto-tune on a track you select the key of the song you're working on so auto-tune can correct the pitch of the voice to that scale then you want to select the type of instrument it is processing so you can have better results there's instrument and bass instrument so you can use it on other things than a voice and finally you have the speed at which auto tune will correct the pitch of each note you can make it sound pretty natural if you keep it rather low but you can also make it sound synthetic if you put it at 0 milliseconds at 0 milliseconds each note will snap instantly to their corrected position so you will miss the natural transition between each note and that's often what is identified as the autotune effect what you're waiting for in only three words that i can only say for if you're looking for a more transparent use of the autotune there are other parameters that are here to help the humanize function will adjust the speed of the effect depending on the speed of the notes so faster notes should be adjusted faster and vice versa with the flextone parameter notes will be corrected only when they are close enough to the target note so this is to keep the expressiveness of the singer what you're waiting for in only three words but i can only say for what you're waiting for in only three words but i can only say four the natural vibrato parameter narrows the range of the vibratos the more you turn it up and that is independent of the pitch correction what you're waiting for in only three words but i can only say for what you're waiting for in only three words that i can only say that's a lot of options all related to peach to manipulate it but see here at the top you have two modes for all of this classic and formant and with formant you have a throat parameter that is similar to the format knob we saw in the vocoder [Music] this one will be a lot quicker to explain the talkbox is an effect that can't really be replicated with a vst i mean there are some but so basically with the talkbox the sound is sent through a plastic tube and this tube goes into your mouth so the sound will be played in your mouth and from there you can articulate words to shape that sound with your mouth it's a bit like when you play music on your phone and place it close to your mouth to make some wawa sound [Music] here it's directly the mouth that will shape formats organically to physically shape the sound it's a very fun effect to use but i can't really demonstrate it as i don't own one myself you can hear a box in songs that i'm not going to play here for copyright reasons but you can hear it on the guitar solo in the middle of the song jambi by tool at 4 minutes and 10 seconds which is one of the best songs in the world or in the intro of bon jovi's living on a prayer for example so that would be all for today so i'll wrap it up here with this video i think i've talked about pretty much everything i wanted to talk about in this series so it looks like it's coming to an end i mean other videos will be added to the sound design playlist but the form may change a little bit from now on and i may do other types of videos in the meantime like more music theories gear reviews and more creative stuff all good fun i actually feel good to finally have all this content available for free for people who might be looking for this kind of information i felt like this kind of thing was missing on youtube as it may not be the most algorithm pleasing content so if you liked this series or found it useful share it around you like share subscribe all this good stuff in the meantime i wish you all a very good day take care and i'll see you all next time [Music] you

Channel: Woochia - Charly Sauret

Views: 754,292

Keywords: music, beat, woochia, beats and bobs, sound design, audio synthesis, sound synthesis, oscillators, modular, synth, synthesizer, video game, lfo, envelope, harmonics, waveform, frequencies, dubstep, FM, AM, fm synthesis, ring modulator, granular synthesis, granular, hard sync, additive, substractive, wavetable, waveshaper, bitcrusher, filter, equalizer, EQ, midi controller, sequencer, step sequencer, vcv rack, ableton, serum, arpeggiator, saturation, distortion, compressor, ott, multiband, mutli band, auto-tune

Id: jWorjBDcty4

Length: 158min 5sec (9485 seconds)

Published: Thu Feb 17 2022