Mixing - Learn EVERYTHING to Start Your Career

1. Introduction: Hi, and the normal producer from start producing music.com. And welcome to my Skillshare mixing class. You will learn everything you need to know about mixing from the tools to the techniques. We will cover it all and practice with exercise files and full mix overviews. This class is one chapter out of my seven chapter course in which I cover the whole music production process from songwriting through arrangement, recording, mixing, mastering, and business practices as well. You will find links to the exercise files in the project section and we'll get assignments as the class develops in the full course, you will also receive multi-track folders of songs from different genres, which you can practice your skills on. Getting under the hood, look into professional production approaches. Start producing music is oriented to help people start their lifelong journey as music producer, whether it's producing your own music or producing others. This program has a lot to offer and you're welcome to go over to start producing music.com to learn more about it. But for now, let's dive to the mixing art form and learn how to stage in the music production process works from beginning to end. I am positive. You will learn a lot from this class. I'll see you inside. 2. What Is Mixing: The mixing process has to end goals first and foremost, it's to make the songs intention clear and beautiful. The second is to deal with technical side effects related to the recording process and technical demands of the platforms that the song is going to be played on later, the first goal might sound obvious, but it is important to note that a great song with a grid arrangement and recording can be completely destroyed by a crappy mix. The mixing stage has a strong artistic impact on the songs since it controls the songs dynamics, tempers, and focal points. The second goal is a bit more complex. People listened to music in many different ways. Ear phones in their cars, phone speakers, laptops. If you want your production to sound great on all these devices, you need to understand their limits and work within these limitations to achieve your goals. So we've already talked about digital volumes, and we know that 0 dBFS is the digital volume limit. That seems relatively simple, but there's a lot more to it since different frequencies sum of two different volumes and our ears receive frequencies differently depending on the volume in which they're played. All these technicalities and many more will be explained in this chapter, I want to point out that understanding the mixing process and experimenting with it, We'll give you a better understanding of arrangement as well since the two go hand in hand. So even if you don't have any desire to become a mixing engineer, I do recommend focusing on this chapter and learning as much as you can about mixing. There will be more assignments in this chapter compared to prior ones. And I will note oftentimes that I recommend using your DAW's stock plug-ins before any fancy third party ones. The reason is that third-party plugins can be very distracting because of their flashy interface and investing some time with the most basic compressor or EQ, will actually make you more focused with the amount of simulations and plugins we have. It's easy to get lost. So I suggest mastering the most basic tools you have and only then expanding your toolbox. That being said, I will present classic tools that are staples of the mixing art since they are important to know as a mixing engineer and a producer, there's a mixing exercise file folder you should download that contains files you can import to a new session and work with according to the assignments. However, before we get into mixing, I'd like to begin with pre mixing, meaning editing, and mixed preparation, because mixing is an artistic process, getting unrelated tasks done beforehand will allow you to become fully engulfed in the process of mixing. 3. EQ pt.1: Equalizers or EQs and short, or tools that balanced the volume of frequencies within a track. They come in handy when there is a need to attenuate unpleasant frequencies, boost pleasant frequencies, or completely alter the timbre of a track. Eq is divide the frequency spectrum into bands or filters. There are three main types of Ben's bell, high or low shelf and high or low cut bell bands are also called peak filters and are used to either boost or attenuate a frequency range surrounding a specific chosen frequency. The queue parameter defines how wide the range around the chosen frequency will be, meaning how many frequencies will be affected from the EQ move you make. The db per octave parameter will define how sharp or smooth the bell-shape will be. Shelves or bands that affect all the frequencies above or below the frequency we chose. A low shelf affects all the lower frequencies, and the high shelf affects all the frequencies above the queue parameter alters the resonance or tilt around the chosen frequency and the db per octave shapes the sharpness of the filter. Low and high cuts are filters that cut all the audio frequencies below or above the determined frequency. These filters are also called low-pass or high-pass, which is a bit confusing at first, but also quite self-explanatory. The low cut or high-pass filter will lead to high frequencies pass, or in other words, cut the low frequencies while the high cut or low-pass filter will let the lowest pass and cut the highest. An important thing to note regarding the low and high good filters is that they do not start the attenuation at the frequency you select, but instead show the spot where there's already a three dB attenuation. The queue parameter is going to shape the resonance of the filter at the frequencies position. And the dB per octave will determine how steep the filter will be. The db per octave parameter was explained in the synthesis segment in the pre-production chapter. But in case you did not see the chapter or completed the chapter awhile ago. Here's a brief overview of frequency is actually a note that when doubled completes an octave, the simplest way to visualize this is through middle a. The middle a is fundamental frequency is 440 hertz, and so the above it is 880 Hertz and the a below it 220. So when we talk about the dB per octave and filters, we are focusing on how much attenuation or boost will occur by the time we reach the frequencies Octave. For example, if I put a low cut filter at 100 hertz with an 18 db per octave slope by 50 hertz, there will be 18 dB attenuation. There are a few more bent bandpass, which will only leave a certain frequency range and tilt, which will affect the high and low frequencies in somewhat of a shelf manner. The last parameter is output gain, since the overall volume of a track will change when we alter the frequencies of it, we will need to level match the process track to the unprocessed track. There are two good reasons for this. The first is to keep our gain staging healthy. And the second and more important one is that it will let us really hear whether the processing we did was good or bad to the source. Remember, louder makes us believe things sound better. So when we AV audio, it's important to make sure the levels are the same to really assess the effect of our work, the frequency spectrum is roughly divided into a few regions, sub bass, which is between 20 to 60 hertz base, between 60 to 250, low mid, between 252500, midrange, between 502 thousand hertz. I'm ends between two thousand and four thousand, presence between four to six K and brilliance from six K upwards. In practice, these terms are used more fluidly in the professional world, meaning that engineers don't focus as much on specific numbers, as much as they do on the energy of frequency range has, let it is important to know them since you will come across them a lot before you move on to the next segment. Do this exercise. 4. EQ pt.1 - Assignment: Open the course exercise files and import the string channels to a session, insert your DAW's stock parametric EQ and experiment as so, we'll start with attenuation. So engage both a high pass filter and a low shelf at 150 Hertz. Try different slope variations on both filters and play around with the low shelf skiing. Notice the different results and options the two filters offer. Ask yourself, does one sound more natural than the other? Does the track sound better at all? And after you've invested a few minutes experimenting, insert a belt filter and see how that differs from the two. Then do the same process with a low-pass filter, high shelf, and a belt filter on the high frequencies. After you're done with that, let's experiment with Shelf boosts and compare them to build filters, place the shelf around ten K and raise five dB at another bell filter at the same frequencies and switch between the two, play around with the bills Q and gain, listening to the subtle differences these two filters have. And then just go wild and experiment however you'd like when you're done with these exercises, import the overhead channel, loop it and go over the same exercises again. 5. EQ pt.2: We want our mixes to translate well in any listening environment, EQs, along with other mixing tools, will help us with that if we use them to create a presence within a frequency range that exists in all listening devices. That range is the mid-range. Mobile speakers and phones don't play very low or very high frequencies. So if you end up mixing elements in a song that only exist in the low or high extremes, they will be inaudible on many devices are good example for this can be a base that has a very loud fundamental frequency compared to its overtones, you can slightly reduce the fundamental frequency range with a shelf filter, which will in turn make your overtones louder when you compensate for it, the gain-loss, this new inner balance will make sure that the instrument will have a stronger presence in the mid-range when it's called for. Another common practice when using an EQ is called sweeping, which is the process of raising a narrow bell filter and searching for frequencies that stick out and need reduction. This is a very useful practice, but it can also be very problematic since having a frequency boosted 20 dB is not going to sound good in any case. Just this, don't go randomly hunting for resonances, but when you do hear a resonant frequencies that bothers you whistle or sing the frequencies pitch. And once you've memorized this pitch and you know what you're looking for, then start hunting. This will prevent you from finding things you're not looking for and from ruining your source with unnecessary notches. These were technical or repair uses of EQ, but in EQ can also be used to flatter tracks. If you find a certain frequency range more appealing than others, you can enhance them and play with the sources timber, for example, you can emphasize the air of vocal performance has with a high shelf filter. So why was her fleur de stride as Eugen, the guided me in this world, enhance a pig's head on a string of an electric guitar or bass. Having the wooden tamper more clear on a drumstick hitting the rim. You can also use EQ for stylistic reasons, like filtering both the high and low frequencies to create a telephone effect. Mixing is a complex puzzle that requires a wide perspective. When you are working on a track, you should always keep in mind what role it plays in the arrangement and how it interacts with the other elements as well. A good way to maintain this focus is to keep yourself from working on tracks soloed. Listening to a track in contexts will reassure that you are not losing your perspective and overprocessing unnecessarily. And other similar problems that EQs solve is called mud, which refers to low-frequency build-ups that result in an unclear mix. Some engineers have low cut on channels that aren't necessarily playing in the low register is just to make sure no low-frequency information will be added from unexpected factors. This is especially relevant when you're working with recordings from home studios. Since something as simple as a truck passing by an air conditioner or a fridge might introduce noises without you noticing cutting frequencies between 30 to 80 hertz will be beneficial in that case. And in some cases you may even want to go higher, but make sure you are not harming the tracks fundamental frequencies as you do so, managing the low-end in a mix is one of the hardest tasks of the mixing engineer, knowing how much or how little to EQ will take time. As a general guideline, My advice is less is more, but be willing to do whatever it takes to get your track where you want it. There's an ongoing discussion whether digital EQs are better off being used to reduce frequencies or excusing them in an additive manner can also prove useful. Using digitally used only in a reductive manner will require a different approach. For example, instead of using a high shelf to enhance the high frequencies of track, you will add a low shelf and reduce all the frequencies until the point you want to emphasize. I encourage you to try both approaches and figure out what sounds and works best for you. But in that case or the other, a good habit to keep up with is level matching. If you've ended up lowering or raising your tracks level, makeup the gain in the output. So you'll be able to AB the processed and unprocessed signal and makes sure that you're helping source and not ruining it. 6. EQ pt.2 - Assignment: As you can see, EQs are a very powerful musical tool that can be used to either help shape or stylized recorded material. In this segments assignment, we'll start with the acoustic guitar channel imported into a new session and give it a listen. This guitar has very strong resonances in the low end when certain notes hit, this needs to be balanced. So open your DAW's stock parametric EQs on it and treat it in whichever way you find suitable. Every EQ move you will make will have price and you'll need to manage the gifts and ticks. I'll give you a small tip and tell you that although these frequencies are picking wildly, I managed to solve the problems with bands that are not attenuating more than 2.5 dB. Next, let's work on stylizing and not unrepaired. Take any channel from the folder and make it beautiful. If it's the guitar, maybe enhance where the peak is hitting and if it's the vocal, maybe the airy frequencies in the performance. If you choose a violin, maybe enhance the wooden tamper or the Bose tamper. First, do so in an additive manner. Figure out which filter will suit the job best. And then when you achieve that goal, try doing this with other filter types. Make sure you are in level matching as you go so you don't fool yourself. And after you're happy with the results you've reached, insert another one right underneath it and try making the trek beautiful with reductive EQ AB the results and see what sounds better to you, what feels more natural and think what was more intuitive for you as well, explored these two approaches and have fun. In the next segment, I'll introduce various kinds of accused and show a few important emulations you will surely come cross in order to help you gain familiarity with the industry's Stapleton. 7. EQ Types + Demonstration: The parameter is mentioned in the prior segment, existing only cues, but not every AQ will give you the option to play around with them. The EQ I was using is called a parametric EQ. Parametric EQs let you determine the center frequency, amplitude q, and sometimes even more parameters, as we just saw with the fab filter pro Q, the more classic form of a parametric EQ, will use knobs, and we'll do the same trick looking a bit different. Another type of EQ is the semi-parametric EQ, which lets you determine the center frequency and amplitude, but not the cue. Oftentimes the Q varies as the gain of the frequency changes, but this is predetermined by the designer. A third type of EQ is the graphic EQ, which will give you a set amount of bands on a set amount of frequencies that you can either boost or attenuate to taste like the semi parametric EQ, the cues are predetermined and either change with the volume manipulations or stay the same dynamic accused react to the incoming audio and don't stay completely static. This feature is great since the effect of an EQ move isn't necessarily right throughout a whole song. So dynamic processing might save a lot of automation work, which we will be talking about in a different segment, linear phase excuse, or a potential solution to a side-effect that sometimes appears from the traditional EQ design. Most of the EQs, you'll meet our minimum phase EQs, which implies that they have slight time delays, which in turn introduce slight phase shifts vary by the frequency, filter type and gain. This is called smearing, and it's more likely to happen when using intrusive EQ moves like very narrow cuts or high-pass, low-pass filters when smearing is not desirable, linear phase EQ is come in handy because they keep everything in time, therefore preventing facing issues. The downside is that they create higher latency than other EQs and might soften transients. So this tool has its virtues and trade-offs. Use it when needed, but know the consequences when it comes to analog excuse or emulations of analog EQs, we need to understand that a big part of the EQ's character comes from its circuitry and design. Some accuse have tubes in their circuitry and give the audio a bit of tubes saturation. Some are solid state and as a result, are very clean, somehow very colorful transistors that gives them their own unique tone. Although modern EQs are practically limitless, classic EQs are still in the forefront of mixing for a good reason. With that in mind, I want to show you a few classic EQs. You will surely see a lot as you dive deeper into mixing. I'll explain their logic since some of them are not too intuitive. But before we start, I want to introduce you to a few basic symbols that will help you get around a new EQ, even if you're not familiar with it this time, marks a bell-shaped filter. This marks a low shelf. This is the locate, this is a high shelf. This is a hiker and this is the phase foot sign. Let's start with the SSL parametric EQ. This is a British EQ that has four bands, which in the default forums or low shelf, high shelf, and to mid-range bell bands. Some emulations will have only one local filter and some will have another hike up filter as well. The two mid-range bell bands have acute knob to them, making the EQ fully parametric. And the high-end low shelves have a button that can turn them to Bell filters instead of shelves with predetermined queue. Here's the SSL inaction. Monday evening. My e evening strolls. My evening strolls, my evening strolls, my evening straw. The next classic EQ is the Neve 1073. This semi-parametric EQ is also British and is very famous for being a part of countless influential records that we are all familiar with. This EQ has high and low shelf Dan's one mid-range bell band and a high pass filter. The frequency of the high shelf is fixed at 12 K. The low shelf has four selectable frequencies. The mid-range bell band has six frequencies you can choose from with a fixed Q. An interesting thing to know about this EQ is that the local filter has a bit of resonance to it, which will add a small boost at the chosen frequency point. This can be used to fatten up a source while still filtering the really low frequencies in other classic Navy Q is the 1081, which has one more bell band in the mid-range, more frequencies for each band to Q, options for the mid bell bands, as well as the option to change the shelf bands to Bell. And finally, a high carb filter. Here are the 10731081 in action. Ms. E evening, Monday evening stromal cells. My evening strolls. My evening strolls. My evening strolls. Ms. E evening. The next dQ is the API, five-sixteenths graphic EQ. This is an American EQ with ten bands jumping in octaves, meaning that the frequencies are doubled from one band to the next. The queue narrows as the boost gets more extreme, giving this EQ, or very musical and pleasant sound. This feature will also be found on APIs, semi-parametric EQs, which are the 55855 dB. You might have noticed that the 550 B has another bat, but the frequency is on the two EQs are also slightly different, so they will be serving different purposes. They both give the option to change the high-end low bands from shelf to belt. And 550 a also has a band pass filter cutting anything below 50 hertz and above 15 k. The frequencies are chosen using the inner blue knob and the gain by the outer white knob. Here are the API five sixty, five fifty in action. The last classical q will be the pool deck. This tube EQ was designed in 1951 and became a classic because of its unique tone and features, the low frequencies can be manipulated with two shelving filters that attenuate and boosts simultaneously, therefore creating a unique filter shape that cannot be achieved otherwise, the next band is bell-shaped and can be used to boost selected frequencies between 316 k. This band also has a bandwidth knob, which is the queue. You can adjust the bell to be very sharp or very broad as desired. Finally, there's a high shelf that can only be used for attenuation. This weird-looking, unintuitive EQ was a big part of some beetles and Motown recordings, as well as countless other historical productions. Here's the pool deck in action. There are many more EQ to learn about, but these few are a good introduction into the analog EQ world. You don't need all these EQ types if you want to start mixing stock plug-ins are generally more than enough. The emulations are more fun to use, but have nothing to do with acquiring skills as a mixer. That being said, they are interesting as they have helped in developing the art form as we know it. So familiarity is useful. Now, I want to present two tests, one for this segment and one concluding the EQ topic. 8. EQ Types - Assignment: First is relevant only if you have an analog emulation in your DAW or if you have any third party plugin, start by importing any track and inserting three different analog emulations and figuring out what bands each offers which frequencies they let you manipulate and which controls you have over the bands, the limitations some analog gear presents is actually what makes them so unique. So try noticing what each has to offer after the interfaces figured out, compare the emulations even if the bands are not fixed to the same frequencies, raised the high shelf three dB and then a, B, three reactions. Then compare the low bands, maybe that as well, the mid bands, the filters, remembering what each analog gear has to offer will make it easier for you to recall it when you need it. The deeper you go, the more you'll know. So after you're done researching the first three, you can do the same with a different batch. The second task will be to try and mix the rhythm section of a song, meaning the drums and bass. Until now, all the EQ tasks you received only include single tracks. But as we mentioned, a mix is a complex puzzle and working in solo is not the way to go. So download one of the songs from the multi-track folders and try to mix the rhythm section of it, whether it's one of the electron pieces or any of the acoustic ones, the rhythm section will have frequencies ranging from the lowest to highest. It's a good place to start with. Notice the low end and start with figuring out whether you're cleaning the base is sub frequencies in order to leave it for the kick or the other way around. Then notice the mid-range. How does the base is meat, so to say, interact with the snare or higher tones of the kick than the highest is the track calling for shiny high hats or relatively dark ones. Notice how altering frequencies affects the feel of the groove and the emotion that the song predicts more than anything. Upfront. 9. Compressors pt.1: Compressors are tools that allow us to alter the dynamic range or the envelope of a track. The term dynamic range refers to the distance in volume between the softest and loudest points of a track and the envelope is the trucks behavior. In other words, compressors help us control the volume of our tracks and can be used for taming and balancing attract so it sits better in the mix or in another manner to exaggerate the dynamics and make things actually stand out. More. Compressors have very few parameters that work in conjunction with each other. These parameters are threshold, which is the level in which the compressor will start the compression process. Attack will determine how much time it will take compressor to start compressing once the track's volume crosses the threshold, release will determine how much time it takes the compressor to reach 0 dB gain reduction after the volume drops below the threshold ratio, which is how much compression will be occurring once it kicks in. For example, a four to one ratio will mean that every four decibels crossing the threshold will only add one dB to the output signal knee, which is how soft or hard the transition into compression will be. Soft knee will result with the ratio gradually rising as the signal reaches the threshold. While hardening will mean that the ratio will just work as it is once the threshold is crossed, output gain compensates for the level loss from the compression process and lets us level match the signal. And finally, side-chain filter, which will tell the compressor which frequencies to ignore when analyzing the incoming audio, fast attack time will flatten our transients, resulting with less dynamics. And slow attack times will accentuate transients. Slow attack and also used to achieve compression that is gentle and transparent. I swear I didn't speak as a monk be spin tonight. I swear I didn't speak as a monk be spin tonight. Fast release times can raise the tracks tail or ambiance and enhance the tracks excitement. That being said, it can introduce distortion and make a bit of a mess if implemented inappropriately. Slow release times can result with more natural sounding compression and tighter sounds, so to say, but can also choke the signal when it is not breathing with the tracks rhythm. Compressors have a detector circuit that listens to the incoming audio and then triggers compression accordingly, the sergeant filter is used to shape the audio arriving to the detector in order to control what triggers compression, different frequencies add up two different volumes and the low frequencies add up to the highest energy and can create a pumping effect that is not pleasurable. So when you are compressing trucks that have low frequency energy, which you don't want to trigger the compressor. You can filter the low frequencies from the detector and have the compressor react only to the higher frequency energy. Let's listen to an example. Notice what happens as I raise the filter. As you probably heard, the pumping cost by the low frequencies is reduced and the overall level the compressor reacts to with lower. This simply makes the compression process more accurate and prevents these unwanted pumping sounds. It's important to note that the compressors parameters all work together and don't really stand by themselves. Let's see this in practice. I will use a very visual compressor to make the principal queer. But remember that when we mix, we work with our ears and not with our eyes. Let's take the vocal track from the mixing exercise files as an example. I swear I didn't speak as the mike be spin tonight. The first thing to do when you want to compress audio is to figure out what it is that you want to achieve. So I hear that the word I and the mic, or pumping out a bit. And I want to balance them out with the rest of the track. As a starting point, set your compressor to an eight to one ratio with a fast attack and release times and a hard knee. Now let's lower the threshold as the trap plays. And here, when the compression starts kicking in, I swear I didn't speak as the mike be spin tonight. I swear I didn't speak as the mic be spin tonight. As we add, didn't speak as some may be spin tonight. I'm taking the threshold further than it's called for. So we can hear the compression in its extreme and dial in the next parameters to meet our needs more easily. Let's start with the attack. We will slowly open up the attack and listen to where it lists the transient or initial peaks, come through and tailor that until we reach an attack. We'd like the sound of a swear. I didn't speak as a monk be spin tonight. I swear I didn't speak as the mike be spin tonight. As we add, didn't speak as they might be spin tonight. After we set our attack time, we'll figure out the release will slow down the release until we hear the compressor moving with our source and not choking its dynamics. Since as I mentioned, the goal we are working towards is a balanced and natural sound, as we add, didn't speak as it might be spin tonight. I swear I didn't speak as the mike be spin tonight. I swear I didn't speak as they might be spin tonight. After having the attack and release that, we can figure out what ratio we want for our compression. Let's start with two to one and slowly raise the ratio to see the effect it has on the source along with how it interacts with the attack and release parameters. I swear I didn't speak as the mike be spin tonight. I swear I didn't speak as the mike be spin tonight. I swear I didn't speak as the mike be spin tonight. I swear I didn't speak as the mike be spin tonight. If your compressor gives you the option to play with the knee parameter, you can set it before or after you adjust the threshold, then it will smooth out the entrance to the full compression ratio and it can help refine the compressors reaction to incoming audio. Now that we have these parameters that we can adjust the threshold, don't be surprised if some settings need to be refined because as I mentioned, all the parameters work in conjunction. And the fact that the threshold is being adjusted might mean that more adjustments will need to be made. I swear I didn't speak as a monk be spin tonight. I swear I didn't speak as the mike be spin tonight. I swear I didn't speak as a monk be spin tonight. I swear I didn't speak as a monk be spin tonight. I swear I didn't speak as the main be spin tonight. The final step will be making up for the level loss with the output gain, compressors will have gain reduction. We use sometimes marked as GR, that will show you how much level is reduced. But I recommend listening to the processed and unprocessed signal and adjusting the volume between the tool since the compression can create different forms of level change. I swear I didn't speak as the mike be spin tonight as well. I didn't speak as the mike be spin tonight. I swear. I didn't speak as the mic be spin tonight. I swear I didn't speak as the main be spin tonight. Setting the compressor in this way is a very good practice to use until you feel you've wrapped your head around the subject. Start with a low threshold and fast attack and release times get the attack set than the release ratio and threshold. Tailor the settings to fit the final threshold point and finally, adjust the output gain to compensate for the level loss. 10. Compressors pt.1 - Assignment: The assignment for this segment will be to explore the compressors parameters and see how they interact. Try compressing the acoustic guitar or bass number to track from the exercise files and compress them with the intention of getting the dynamic steady. Use your stock compressor and try reaching around five dB of compression in various ways. Try having a fast attack with a very high threshold that might sound good, maybe lower the threshold but slowed down the attack. See how the ratio affects both settings. Notice that changes you make will require some adaptations on other parameters as well. And make sure that the truck is still breathing and does not feel choked. After a few minutes of experimentation, tried doing corrective EQ before the compressor and see how that changes the way it behaves. Altering the harmonic balance will also change the dynamics in certain scenarios. Experiment with this and see how it appears in practice. In the next segment, we will further explain how to use the compressor and hear how it can be used artistically. 11. Compressors pt.2: The last segment, the parameters of the compressor were explained and we used compression to control the level of a track. In this segment, I want to show and explain how compressors can help shape the envelope, character and tone as well. For an example, this is the same track with two extreme examples just to make a point. But although as I mentioned, all the parameters work in tandem and can't really stand by themselves. It's safe to say that the attack and release parameters will be the main influence on the tone by the compressor. Just again, perspective, you should know that in the analog world, fast attack is around 20 microseconds and faster release is considered to be around 50 milliseconds. Compressors with ten millisecond attack times are considered relatively slow and release times can cross the 1 second range. Digital compressors can give you even faster and slower at times to work with if you need them, fast release times, we'll bring excitement and attitude to attract while slow-release times will keep your track more tight and smooth. Fast release times might help in unstable performance can be used to change the timbre of a track or just be a problem. So be mindful of this parameter. Slow-release will mean keeping the compression going for awhile, which can help clean the track and smooth out the next transient. But you'll need to make sure it reads with the tracks groove because it can also choke the sound if it's too long. So again, be aware of how your release is affecting the audio and notice if it serves, your intentions are not. The same principle goes for attack times. Relatively fast will result with aggressive tones, while slow tax will be more natural and smooth sounding like EQs compressors can be used for either pair work or in a stylistic manner, adding character and tone to tracks. Let's take a look at a few examples and see how they come to play. Here's an acoustic guitar. Let's say I'd want to accentuate it's transients and introduce more attitude of They'll just enough attack to let the transients through before the compression kicks in and set a fast release. So the compressor will get back to 0 dB gain reduction before the next trend incomes in beside the dynamic effect the compression had on the track. Notice what it does to the tracks character. It feels more energetic and it puts it more forward. Here's the snare track. In this case, I want the ghost notes to be more audible because they are really important for the group. So I will set a threshold of the compressor to attenuate the loud hits and I'll fast released so the ghost notes will be exempt from the compression. Next example is the string track. I want this section more balanced but still dynamic. So I'll dial in a relatively slow attack and medium-term fast release to have it in place, but with the compression not to Audible. Notice the section sounds more glued together, so to say, since the dynamics work together, although digital emulations do manage to bring the character of classic analog compressors. They still do react a bit differently to audio, especially when there's a lot of gain reduction going on. So when working in the digital realm, sometimes engineers chain several compressors, one after the other. Each will be compressing just a bit for the final effect does sound smooth. So as you can see, compressors to or a very musical tool and have emotional impact as much as any other tool we use. It is multipurpose as it can be used for dynamic control, Somewhat of distortion, even EQ in a way and to enhance emotion when it's needed. 12. Compressors pt.2 - Assignment: This segment assignment will be to compress artistically, take the tracks I've used in this segment and experiment with them yourself. Try and making them more aggressive, and then try making them feel softer. Be aware of the emotional impact. Technical details have an after you're done with the guitar, strings and snare, move on to the other source material and the exercise folder. In the next segment, we'll go through the various types of compressors and the differences between them. 13. Compressor Types: The last thing to get familiar with before with some compression topic is the various types of compressors, since some are not as intuitive as others, I will show and explain how to use a fuel and hopefully save you some confusion when you encounter them yourself. Digital compressors can usually be fully tweaked. They are also very clean, making them a good tool for anything. Practically the compressor I used in the demonstrations in the prior segment is my DAWs stock plugin. But there are many third-party plugins that have some very useful features incorporated. Fet stands for field effect transistor. These compressors can reach very fast attack time because of their transistor circuitry and can be used for anything from vocals, Bayes, rule max, or anything else that calls for fast and clean attack settings. The 1176 is the most famous FAT compressor and you will probably come across many emulations of this classic mixing tool. The 1176 has a fixed threshold level. So in order to start compressing, you need to push the sources level with the input knob into the threshold and then balanced the volume with the output knob. The 1176 is attack ranges from 20 microseconds to 800 microseconds to release ranges from 50 milliseconds to 1.1 seconds. And an important thing to know is that the attack and release parameters are counter-intuitive. The highest number represents the fastest attack and the lowest number represents the slowest. The last cool feature of the 1176 is the all buttons in mode, also known as British mode, by default, when one ratio was chosen on the compressor, ratio chosen before is de-selected. But British engineers discovered that all the ratios are pushed in the compressors reaction is mild distortion and extreme attack and release times. This gives a lot of character and can be used to give a tracking extra pop of excitement. Optical compressors have an interesting design. The audio going through the compressor passes through a light element that lights up and down as the input level changes surrounding the element, there's an optical cell that attenuates the audio as delight grows stronger, these compressors have slow attack and release times and are considered soft and warm. Optical compressors are good for sources that need slow attenuation like certain vocals, base or string sections. L2 is the most famous compressor and also has many emulations. You will surely come across DLA to a has two naughts, peak reduction, which acts like a threshold knob and gain, which is the output level. Generally speaking, the LA two-way has a soft knee in average ratio of 41, average attack of ten milliseconds. And to release stages, 50 per cent of the release happens around 60 milliseconds. And the second 50 per cent can take between one to 15 seconds. But all of these parameters I just mentioned actually fluctuate according to the audio coming through the compressor. So as you can see, although it is very simple to use, the LQA is actually a very complex and musical compressor. Tube compressors lives and their circuit and create the desired warmth that is considered the signature tube sound. These compressors will be great for certain vocals, string section, and any source material that can use tube tones. The most famous compressor in this category is probably the Fairchild, which has 20 tubes in it. This compressor has an input, a threshold knob, and what the designer called time constants, which are different combinations of attack and release parameters. The tech times will range from 200 to 800 microseconds and release times will range from 300 microseconds to 25 seconds. In extreme cases, if you purchase in emulation of the Fairchild compressor, makes sure you read the manual. Since this compressor is very unique and has a lot to offer, VCA stands for voltage controlled amplifier. The way these compressors are built gives them the ability to be both very fast and slow. Because of this flexibility, they are often used on the master bus or on drums with the best example being the SSL bus compressor, which gives a wide variety of attack and release times are different. Famous VC compressor is the dvx 1 sixth, which is actually very limiting in terms of options. Since the SSL is more flexible, it will oftentimes be used on the master bus. Where's the dp? Dx is a classic punch enhancer for drums because of its very fast and aggressive attack. Multiband compressors are actually multiple compressors in one unit that are divided across the frequency range. This gives us the option to compress specific ranges differently, or to compress only one specific range instead of the whole track, there are analog multiband compressors, but they aren't as commonly emulated. So you'll mostly see digital compressors like the waves C4 or C6 and the Fed filter Pro and B. If you have any of these emulations, go ahead and experiment with them. Having a wide variety of gives you a lot of creative freedom, but also a lot to learn. The way I suggest you go about learning different types of compressors is to choose one compressor and try using only that one to reach your compression goals. Do that at least for awhile, and learn their capabilities both as a dynamic tool and as an emotional enhancer. After a few days or a week of consistent use, change to a different compressor and learn that one thoroughly. Doing this will force you to experiment and know your tools in greater depth than just randomly picking a few and working with them. 14. Expanders / Gates: The parameters you'll meet in this segment are the same as in the compressors, one added parameter range as opposed to compressors expanders, or actually used to increase the dynamic range by making the quiet parts of a track even lower. The audio above the threshold stays the same, and the audio below it will be attenuated according to what you have dialed in the range and ratio parameters. The range parameter will limit the amount of attenuation the expander introduces. And the ratio will define how much attenuation will occur. The attack and release times define how fast or slow the expander of works, just like in a compressor. Here's an example of an expander in action. Noise gates are simply expanders with extreme settings that attenuate the audio under the threshold to silence, the range and ratio parameters are flattened out and then the attack and release parameters are used to tailor the effect and make it sound natural. Noise gates are mostly used to clean up tracks, which is very useful because when a track is heavily processed, excess noise in the track will be enhanced. And so having a cleaner track allows you to process the audio as much as you pleased without the unfortunate effect of enhancing unwanted noise. The noise gate on close magnetic drums is very useful because it can give you more control and have a channel functioning fully as a drum channel with no effect on how loud other drums are. All that being said, it is important to note that sometimes having ambient noise and Mike bleed from multi-model instruments is actually helpful. So don't just use gates for the sake of cleaner trucks. Notice whether its effect is helping or not and use Accordingly. I personally don't find myself using expanders much, but I do find noise gates quite useful in mind mixing, experiment with these tools and figure out what works for you. The essays or compressors that focus in the high range of the frequency spectrum and are used to control siblings S sounds and chew sounds can pop out in tracks and be somewhat uncomfortable in a way and the ester is come to solve this problem when you work with the DSR, The first thing you need to find is where the frequencies you are. This will change from singer, singer and from Dr. Mike and will also vary depending on the continent. You might need one ds or for the essays and another for the two sounds more so sometimes having to the ester is sharing the burden over the same siblings will sound more natural than having one working alone. Once you've found the frequency range you want the desert to work at, you need to define the range, which is the maximum attenuation the d'Azur will have to work with. Some of the hazards will stop there, and some will have more parameters to play with the threshold or even more band types for detecting the siblings in the audio, you don't want to overdo yes, since this might create a sensation similar to lists being which might upset your singer. I swear I didn't speak as the mike be spin tonight. So although they might sound simple, it should be dealt with with delicacy. Dsrs can also be used to reduce resonances in tracks that aren't. Vocals can be used on guitars, strings, or any channel that has high frequency resonance is coming and going. Limiters are compressors with very high ratios that will limit the audios dynamics tend to one ratios and above are already considered limiting. And there are limiters working in 100 to one ratios and up to infinity to one. These limiters are called brickwall limiters and are used to reduce the peak level and therefore enhance overall level. If what I just said confuses you, go back to the level and metering type segment in the recording theory chapter. But in short, the overall level of a track rises as the dynamic range is reduced. Imagine that your audio is being pushed towards the ceiling. The quietest part of your audio rise as you push the level upwards. And the first thing to reach is the peaks. The more you push, the louder the audio will get at the expense of your dynamic range, since the peaks have already reached their limits and there are now restricted by the ceiling. Notice how the LU AFS levels rise as I pushed the audio more into the limiter. It's important to note that brick wall limiters alter the wave forms because of the extreme compression and they might cause digital distortion that will be audible. Generally speaking, you should be careful with these tools because they are very extreme. Limiters are often used on the master bus at the final mastering stage, but can also be used on specific tracks that need their peaks reduced, like acoustic guitars or drunks. 15. Expanders / Gates - Assignment: Takes nerve number three and try expanding and getting it yourself. If you find yourself confused, get back to the beginning of the segment and watch again, but try to get your head around it logically. Then the S, the vocal track, the singers, siblings is very strong and needs taming, but makes sure you don't overdo it and have her sound as if she's listing. Lastly, take one of the mixes from the mixins folder and limit them. See what happens when you push too hard, play around with the attack and release times and see how that affects the limiters. And then try reaching a result that is actually true. 16. Saturation: Saturation is referring to either mild or extreme forms of distortion and is one of the mixers most creative tools. Saturation can be achieved by overloading a tube circuit, transistor circuit, tape prehaps petals and through many other ways. In essence, distortion is altering a waveform and changing its harmonic content. And it can be used on any source to any degree for mild tonal shaping or complete destruction. Many of these circuits or forums of distortion were modeled in order to have these temporal manipulations in the box, meaning and the computer. And so even in today's clean digital days, we can add some analog sounding tones even when working with clean modern equipment. Let's start with a bit of theory and then get to the practical sounds and uses of saturation. Saturation will add frequencies, or in other words, harmonics to your signal. There are even order harmonics and odd order harmonics, sometimes regarded as second, third order harmonics. Even order harmonics will be even multiples of a frequency. For example, if we have a sine wave which has a single frequency playing at 100 hertz. Saturating it with even order harmonics will create a frequency in 200 hertz than 400, 600, and so on. Odd order harmonics will be multiples of odd numbers. So the overtones created from the saturation will be three hundred, five hundred, seven hundred and so on. Since most of the audio files you'll be using, we'll have way more than one frequency. The calculations will be far more complex, but the math isn't as important as understanding that saturating or distorting the track will create and change frequencies in the track you're altering. Even order harmonics are sometimes regarded as more pleasant than odd order harmonics. I personally don't give a **** and recommend experimenting and using them both. Now let's get the practicalities. Here's a clean vocal. I swear I didn't speak as the mike be spin tonight. Here's the same vocal with mild saturation. I swear I didn't speak as the mike be spin tonight. And here it is with extreme saturation. I swear I didn't speak as a monk be spin to me. Let's see the waveforms of these three examples. As you see the peak information and the shape of the waveform, or in other words, the dynamics and harmonic content of the track, I've completely changed. Now let's listen to different kinds of saturation and try and noticing the difference in tone. This is tube saturation. I swear I didn't speak as in mind, me spin to me. This is tape saturation. This is transformer saturation. I swear I didn't speak as it might be spin tonight. Hearing these delicate variations between the different saturated might take awhile. But when you train your ear and use them tastefully, you can reach tones that aren't reachable with either EQs or compressors, since saturation does not only manipulate existing frequencies, but also create new ones, saturation also alters dynamics and can be used to control peaks and raise the overall level of tracks. For example, here's the snare. So this is where the peak levels are. But notice what happens when I mildly distorted. The peak levels are lower, but the volume feels the same. Distortion can either be used as a stylistic timber shaping tool or as an advanced EQ or compressor more so it's a lot of fun. So take your time and play around with distortion and learn how and when. This can be used to achieve tonal goals that EQs and compressors can't. 17. Saturation - Assignment: Open a distortion or saturation plug-in you have in your DAW on the snare number one track, the acoustic guitar track and the vocal track from the exercise files start by lightly distorting them and notice what happens to their peak levels and tone increase the saturation gradually. And notice what happens to the ambiance or Mike lead the transients, how clear the information is and when you reach a point in which the truck is completely destroyed, stop and back the distortion off a bit after this slow experiment phase at an EQ, before the distortion plugged in at a belt filter and start moving it around. Slowly, raise it gradually in the low frequencies and see how the saturate or reacts. Then raise it in the mids and see again, then raise it wildly. Since distortion will add harmonics according to the incoming audio, raising low frequencies will result with added high frequencies as well. The final experiment is to add an EQ after the saturation and see what manipulation this offers. Have fun. 18. Channel Strips: Channel strips are simply a channel from a console. They usually consist of a preamp, a dynamic section consisting of an expander OR gate and a compressor and an EQ, having everything laid out in one strip is very simple and useful regarding workflow. And even nowadays we have channel strips emulated from specific consoles that model each component and give their character. Modern plug-in companies even have modular channel strips in which you can customize and choose each module and build yourself your own custom channel strip. So the SSL channel strip looks like this. And you can see that we have a filter section here, the dynamic section here. This is the compressor in which you can set the ratio threshold and release. This button will control the attack and will either be set fast attack or a slow attack. And this is the expander OR gate which you can set using these three knobs. Here you have the famous SSLD q, consisting of low and high shelf that can be changed to Bell filters and to mid-range bell bands. As you can see here, you can either take the dynamic section in or out and the compressor or expanders specifically in or out. And here you have the option to engage or disengage the EQ, the dynamic side chain button. We'll use the EQ that you have created in this section and send it to the side chain circuitry or more accurately, the detector circuit of the compressor. If what I just said was not clear for you, I have a video covering compression in depth in which I explain what side chain and the detector circuit in the compressor. Or lastly, the pre dynamic button or change the signal flow inside the channel strip and place the EQ before the dynamic section. As a default, the signal flow goes as so. You have the preamp, the inputs going to the dynamic section, that then goes to the EQ section, then to the filter section, and finally to the output fader. This will simply mean that the audio goes from the pre-empt to the EQ. And then dynamic section. Let's now take a look at a different channel strip by Lindell audio. And let's see how the logic that we just learned from the SSL comes to play with this plugin. Here we have the preamps section, which has an input and output knob that we can use for saturation, a high boost or low boost and high-pass, low-pass filter. Next up we have the compressor. This is in 1176. It also has an input and output, not the option to play around with slow, medium and fast attack and release times. And the dry wet knob as well. This compressor two has a sergeant filter which you can learn about in my previous video, a few ratios to choose from, and the option to decide how much each side of the stereo effects the other. Lastly, we have a protocol AQ, which you can also learn about in my EQ types Explained video. And you have the option to play with the signal flow using these arrows. Now that we understand the logic, let's play around and see how this comes to action. Are we using a loop from my upcoming sample pack which you will also receive first if you sign up to my email list. So this is the loop unprocessed. Let's start playing around with it. All. Disengage the compressor and the acute. And we'll start with the preamps section. I can go from mild saturation to pretty extreme saturation if I want. Now, let's move on to the compressor. Moving on to the EQ. So as you can see, having everything set in one package is very useful for workflow. Let's look at a last channel strip just to get the information embedded in our brains. This is the API channel strip by UID and the signal flow just goes from top to bottom and from left to right, you'll have the pre-empt section going through the filter's going to the expander gate, then to the compressor, and then to the EQ, and lastly the output fader. So here you have the input section consisting of a mic and line input that you can change here, you can saturate your audio using these knobs. You have a pad attenuating the level of your audio and a low cut filter. And lastly, a phase flip button, a low-pass, high-pass filter that you can engage here and sent to the side chain as well. Then the classic dynamic section, you have the gate expander here. You have these knobs on the compressor changing your knee from hard to soft and the compression from new to old that if I recall correctly, changes the compression type from upward compression to downward compression. And cool feature that you inserted here is that you can change from the five fifty, five sixty EQ here you also have the predynastic button, which will change the signal flow and the dynamic side-chain. So although they might seem unintuitive in the end of the day, channel strips are pretty simple to get your head around. So the next time you see a channel strip, don't be daunted or confused. Just try to understand the signal flow or to engage and disengage modules inside the strip. And to understand if the channel strip gives you any special features that are unique to it. 19. Reverb pt.1: Reverberations are numerous reflections of a sound that reaches our ears after meeting one or many surfaces, the length and timbre of these reflections give us an indication of the size of the space. The sound is n, and the material the space is built from reverb units gives us the option to create a sense of space around tracks we mixed by emulating how spaces and materials react to sound. There are many ways to create reverberations, and each way has its own unique tone and character. Since recording in a cathedral or Concert Hall is a bit of a hassle, studios found interesting solutions before digital reverbs were invented. A reverb chamber was the first, was simply a room built from concrete or any other reflective material that had a speaker and microphone is in it. Audio was sent to the speaker, diffused in the room due to it's reflective nature and recorded through the microphones. Famous studios had chambers that gave them their iconic sound. And nowadays, many of these chambers are emulated in plugins and digital hardware will hide as we'll go in. Plate reverbs or another solution created by big yet thin sheets of metal. A driver vibrated the plane according to the audio that was sent to it. And the vibration was then recorded with two pickups with spring reverbs use the same approach, but instead of vibrating a metal plate, it's a metal spring which inevitably creates a distinctively different tones. These reverbs have a stylistic stamp on rock guitar history and are still built as part of many guitar amps and petals. Digital reverbs appeared in the seventies and we're programmed to create a sense of space through delaying the incoming signal with many fast delays and filters similar to the reflection of a room. These did not try to emulate a specific room. So different algorithms were given different characters and we're not focused on sounding real as much as pleasing or flattering. Nowadays, we have the computer power to actually model how rooms reverberate and so convolution reverbs were invented. These reverbs use impulse responses, which are audio profiles of a space created by algorithms that analyze how the space reacts to audio will go in and choose Go. Modern modelling for both plug-in and hardware companies give us a wide variety of reverb types to choose from, from concert halls and arenas, two bathrooms and kitchens. Most of the reverb types can be used with long or short the case. But naturally, Paul's caves and chambers will be longer than rooms. For example, the parameters you'll find in most reverbs will be the pre-delay, which is the time that a unit takes before it starts the reverberation process decay or time, which will dictate how long the riverbeds and a dry, wet or mix knob that will allow you to blend the original signal, which is the dry one, with the reverb, which is the wet signal. The pre-delay parameter is useful when you want to create a distinction between the source and the river. Here's a source sent to a reverb with no pre-delay, will hide as we'll go inside and true. But go Dan law now with some pre-delay, will hide as we'll go inside and true. But our goal, then, the vocal is more distinguishable this way, since the reverb isn't smearing the initial audio source, this is something I might want in certain cases and might not in others depending on the style of music I'm working on and the source I'm working with, the decay time will be defined by the goal I'm trying to achieve. As a rule of thumb, long river of times will often create a sense of distance and short reverb times we'll create a sense of space, will go and chew. We'll go inside and to go. Roughly speaking, the case below 1 second are considered short. And as you saw when the short reverb was engaged, it really put the source in space. The long reverb had a similar effect but emphasized Granger more than the field of a specific location. Sends and returns will appear in another segment. But I'll mention in short that the dry wet knob, which might appear as mix, is used when a reverb is inserted on a track and not sent to it when the river is on an auxiliary track, it should be 100% wet. The balance between the dry and wet signal might have an effect on how long you will want your reverb if you decide to have the reverb relatively loud and apparent in your mix, you might want to shorten your decay time so the source won't smear out. There's no right or wrong as much as there's intention. The genre, your work in, your own personal taste will dictate the decisions you make. The only thing I do want to point out is, again, mixing is a puzzle and changes in one place might require adaptations in another other parameters you might meet and reverbs are size, which changes the size of the program, the room, and creates variations in both tone and the K of the reverb diffusion, which changes how reflective the space you've programmed is. And therefore how the fused, the reverb will be early reflection, which changes how far and or loud the initial reflections will be compared to the rest of the reverb BQ filters, usually high or low shelf or cuts that let you either further shape the tone of the reverb or take out frequencies that might be clashing with other elements in the mix. Every manufacturer might add his own personal additions. So look things up in the manual and Casey see anything you're not familiar with. 20. Reverb pt.1 - Assignment: Today's assignment will be to gain some familiarity with a different reverb types. See what reverb plugins you have in your DAW and insert them on the overhead channel from the exercise folder, set them all to 20% wet or so and listen to them one-by-one, noticing the different textures and dimensions they create, then add 20 millisecond pre-delay, AB that to the 0 millisecond pre-delay and try noticing what impact it has or the transients more clear, Is there any difference at all when you're done with that, set the pre-delay back to 0 and change the decay time to 1.5 seconds, go through the reverb types again with the same listening guidelines, and then change the pre-delay to 20 milliseconds and listen again. The third task is listening to the reverbs and a being them with the dry signal. This will really focus your ear on how the reverb is actually changing the source. After you're done with this, go ahead and experiment with different amounts of dry wet percentages, the k times and privilege to deepen your understanding. 21. Reverb pt.2: Reverbs might come across as simple, but there's more to them than just setting the killings and simple practicalities that can be an important part of your arrangement in a way and create some links that will take your listener on an emotional journey. As an example, here's the difference between a dry transition in a song and a carefully processed one. Different types of reverb have varying tonal textures like we saw in the last segment, and can be used to achieve a wide variety of goals. As is the case with most techniques we've discussed, you will form your own approach as you will dive deeper into music production or the mixing world. But as for now, try this. Look at your production or mixed like a theater play. Understand who is the main character or the supporting characters and what is the scenery or setting of the performance? Reverb will be helpful in creating the scenery. Let's have a look at an arrangement and see how this approach comes to practice. Let's get out of this town. Seasonal birds, the tar is threatening to drown. Come on, get up. The song starts with the ping sound on the right acting as the main character. The intro includes a melodic loop on the right and various percussive elements that are panned differently and are sent to a room reverb in different levels. In the intro, the melodic loop seems to be the main character, but when the vocals base and roads come in, the listener begins to understand that it's actually a part of the scenery. Let's get out of this town. Seasonal bird, while the vocal is the main character and the base and roads or secondary characteristics, the vocal is central, reverb very slightly to create the feeling that it's a part of the room but closest to the listener will take anything that feels like home. And the base and roads are not sent to the reverb at all since they're low range does not overpower the sense of proximity that the vocal has. As you see, my focus is what feelings or sensations the tones create rather than what tool is used to try to understand what experience do you want to create for your listener and slowly find a way there in one way or another. 22. Reverb pt.2 - Assignment: As a practice and as today's assignment, choose a few songs or an album you like, and pay attention to how reverb is used in them. How long or short the reverb is, how much reverb is there at all, and what scenery does it put you in? These questions will help you understand why these choices were made. And after you've listened to a few songs with these questions in mind, try to experiment with reverb yourself using the tracks from the exercise folder. Here are a few bullet points to guide your experimentation. See the effect minimal amounts of reverb has over the tone of a track. Notice adaptations you might need to do when you change the amount of reverb that is introduced. For example, if I make a track more dry, maybe I'll need to make the decay longer. Try hearing how E queuing reverb effects the reverb and the source. 23. Delay: Delay or echo, is an effect that repeats the audio coming through it, but don't let it simplicity fool you because this is a very powerful tool that will assist in creating space style and interesting textures in your mixes. There are two basic parameters to consider in a delay. Firstly, delay time, which defines how much time there is between the original signal and the echo. And secondly, feedback, which will determine how many repeats the delay will have. Most delay plugins will give you the option to sync your delay to the project tempo and easily get the echoing effect grooving with the song. You can also do this manually and set the delay by milliseconds. The delay will be audible as a separate piece of audio from 30 or 40 milliseconds onwards. When you use lower delay times, there will be a phase shift because the two waveforms are so close and a chorusing effect will be introduced. I swear I didn't speak as the mike be spin tonight. I swear I didn't speak as it might be spin tonight. The feedback knob essentially sends the output of the unit back to the input and feed it back through the delay circuit, therefore adding another repeat to the effect. This parameter might be called repeats, and it can be used mildly to varying degrees depending on the effect you are trying to create. Different delays will give you different ways to tailor the sound of the delay, like IN locate filters, Q, reverb, saturation and modulation, which will be explained in the next segment. These are used to sculpt the tone or texture of the delay and help you create a distinction between the different ambiance as you create different delays, have a few configurations and we'll use a cuboid by sound toys, which has many. You can set it to a single delay, which is one mono channel, repeating your audio tool, or stereo delay, which will be two channels panned left and right and configured separately to play with your delay ping-pong, which is also a two-channel configuration that is first sent to the left and then sent to the red channel, creating a ping-ponging effect between the two speakers and rhythmical, which will react to a rhythm that you predetermined before digital delays came to the audio world, analog tape was used to delay a signal sent to it using tape gave a certain timber to the delayed signal that is still sought after to this day, units like the duplex and Roland space eco are used quite often in modern productions and are emulated by many plug-in companies. In other old-fashioned delay unit is the Cooper time q, which actually has a garden hose in it that delays the signal. There's two has a very iconic sound as well and is still being used on many records. Then digital delay units arrived in the form of guitar pedals and many plugins, some of them are unique designs and some of them model specific units with their tone and features. Delays can be used in a few ways to stylize, to create space, or as in effect, a good example for a stylistic use of a delay is a trick from the sixties called slapback or slap delay. Slap delays have a single repeat with a fast delay time between 4120 milliseconds that was often used on guitars and vocal and is still used to this day when you want to create a nostalgic reference in productions. Oftentimes delays are used to create a sense of space in a different manner than reverb. For example, here's a dry vocal and true. But our goal then, here's the vocal sent to a ping pong delay will hide as we'll go inside and true. But our goal then it might feel a bit unreal ones soloed, but in the mix, it can be used mildly to give a vocal or larger than life presence. Glaze can also be used to create a few effects that we'll emphasize certain parts of performance. It can also add dynamics and groove through your mix and create beautiful chaos when you feel that's needed. The first example I'll be showing you is a technique called delay throws. After you set your delay, you can send audio to it when you want a phrase to repeat by automation will hide as we'll go inside and true. But our goal then, automation will be explained in a segment of its own. But for the sake of this example, let's consider automation as the process of changing a certain parameter during the course of a song. For example, how much a channel is being sent to a delay. So you can use a few delays for delay throws. Here's an example that's a bit over the top. Just for the sake of demonstration, we will hide as we'll go inside and true. But our goal. Another effect comes from a side-effect that tape delays had when you would play around with the speed of the tape. This might sound familiar and the processes simply playing with the speed of the delay once audio is coming through it, a 3D effect you can create as dialing in extreme amount of feedback until the delay reaches complete chaos, we will go in the tree. When you combine that with the speed variations, you can create really cool tones and have a lot of fun. I'll present the last effect you can create with delays, and it's called the Haas effect, which uses a psychoacoustic phenomenon recording how our brains localized sound. Let me demonstrate here what happens when I take a mono Source, insert a stereo delay on it, and delay one side by only ten milliseconds as we'll go inside. And this technique is mainly used to make a mono or stereo, but this can also be used on stereo sources to create a widening effect and a slight temporal shift. The last thing to talk about now that we're familiar with how a delay works is how to set it up in our mixes. You need to have some level going through your delay in order to set the tone of it. So start by sending a healthy level and noticing these few things. The rhythm of the delay has to work with the groove of the instrument you are affecting and with the songs groove. Notice if the delay makes the mixed clogged and muddy, this can be solved by filtering the low frequencies of the delay or by lowering the feedback parameter. Many repetitions, then the ambiance created from the delay must suit the scenery and emotional effect that you're trying to create in your mix. Just like mentioned in the reverb segment, after you've got the settings dialed in, I recommend lowering the send fader to minus infinity and start sending slowly rum their upwards until you begin hearing the delay. If you're using the delay just to create ambiance, that might be the place to stop. But if you want the delay to have a more dominant effect, go further and just make sure that the track is not overpowered when you're using the delay for the feedback in effect, just take note that you are not pushing the feedback too long or too far because it will just become unpleasant. Play around with both the feedback knob to keep it flowing pleasantly. The time knob to create the pitch variations. Here's an example, so you can see this in action. 24. Delay - Assignment: Open your exercise project and insert your stock delay plug-in on the acoustic guitar channel. Have the plugin with minimal feedback, making it 20% width, set the delay time to 160 milliseconds and filter the high frequencies from ten k onwards. Now play the track and AB the source with and without the delay. Experiment with this slap delay for a few minutes and play around with the high filter, the dry wet balance, and the feedback to see how the two effects the delay, insert a ping-pong delay on the vocal track, set the project tempo to 90 ppm and the delay time to both the right ping and the left Pong to quarter notes set the mix knob, however you please listen to the track with it engaged. Then change the delay time 2.5 notes and decide which works better with vocals group after that is figured out, filter the high frequencies and see if it works better, I really recommend following this practice step by step because it will show you why and how this supposedly simple tool should be thought out after you're done with these practices, you should try and use the techniques mentioned in this segment because they are a lot of fun and can be very useful. 25. Modulation: The term modulation refers to a group of effects that have a modulating parameter in them. You might have heard about phasers, flangers, choruses, trembles, and vibratos. And in this segment I will explain how they are produced and how they are useful. Courses duplicate a signal delayed by five to ten milliseconds and played over the original signal. The proximity between the two waveforms create phasing and therefore frequency cancellation called comb filtering, because of the shape the cancellation has. Lastly, the delay time of the duplicated signal is modulated by an LFO, which creates two things. The first is another variation in the phase relationship, and the second is a sense of pitch variation because of a phenomenon called the Doppler effect. The best example for the Doppler effect is the changing pitch of an ambulance siren as it passes you and drives away. The pitch variation comes from altering the distance or speed of sound wave for a listener, the main parameters on the course will be the rate which controls the speed of the LFO that causes the pitch modulation and depth or intensity, which is practically a dry, wet knob that will control how dominant the effect is. This effect is very popular and useful in a few manners. Firstly, as a noticeable effect on sources like guitars or keyboards. And when it's in a stereo configuration, mild use will create a psychoacoustic effect that makes things sound wider. And additionally, it can also be used on a mono source to create a sense of stereo. Will hide as we'll go inside and true, but go Dan law. Flanges also create comb filtering by duplicating the incoming signal, but use shorter delay times that cause greater frequency cancellation. And a very distinctive tom flanging will affect the higher frequencies more dramatically than the lower frequencies, while coursing will be a bit more noticeable in the lower ranges, flanging used to be created by placing two audio signals through a tape machine and sending it to other machines. One of these two secondary machines was slowed down by placing a finger on the tape reels edge, which is called a flange. Then the two signals were summed to another machine and played back with the effect. There's an additional parameter to the rate and depth on flangers, which is feedback. This will send the effective output signal back to the input, creating a deeper flanging and unique metallic tone will hide as will go inside. Oh, phasers process the audio in exactly the same manner as flanges, but at a few all pass filters, these filters do not change the volume of the frequencies, their position that, but to alter the phase around these chosen frequencies, the rate, depth, and feedback controls will be the same. And some phasors might have an additional pole or stage knob, which determines how many filters the phasor has with the brand. The last two modulation effects differ from the first three because they don't manipulate the audio by duplication and phase alteration, but by affecting the source as it is, tremolo alters the sources volume up and down, and vibrato alters the audios pitch up and down. These effects are classically used on guitars, but can be very cool on anything practically will hide as we'll go inside and true. But go Dan law. 26. Modulation - Assignment: Insert the modulation effects one-by-one on the electric guitar channel and explore them dry, seeing what they add when they're mild, and then what they add when they're extreme. After you're done with the electric guitar, try them on the vocal track, the overhead track, and the bass track as well. If after these you have the energy experiment with all the files you have at the exercise folder. 27. Automation: Automation was mentioned before as changing our parameter with the course of the song. It can be used to refine volumes, sends, and various settings on channels and plugins. And we'll add dynamics and depth to your mixes. Automations can either be performed with the track or programmed with the cursor. Each DAW will have its own way to write automation. I will be demonstrating useful approaches and techniques for automation on Pro Tools that you can then use in any software. Volume, automation can come in handy in a few ways. The first and obvious one is changing the overall level of audio tracks. Certain instruments might need volume adaptations in specific parts of the song. This can be done in a broad way by changing the volume of the whole track. And in a more defined way where you can really go bar by bar and change the volume up or down. This lets you emphasize or attenuate melodic lines that aren't coming through or cutting the mix too much like certain words or breaths in the vocal track and affects returns when you want the extreme transitions. Effects like delays and reverbs can either be inserted on a track or on an auxiliary or effects track that have audio sent to them through sentence. Having the effect on an auxiliary track gives you the option to use one plug-in on many trucks there for saving a lot of CPU power. The volume sent from the audio track is called the sand and the volume on the effects track is called the return. Effects don't need to stay static throughout the whole song and can be automated to fit the dynamics and the emotion the arrangement is trying to create. Here are a few examples of how send automations can be used in the mix. The ability to automate parameters on plugins opens a huge door for creativity as much as it helps with organization, since it saves you from the need to open new plug-ins or duplicate tracks to process them differently. Here are a few examples of that in action. If you want to mute the track, you can either lower its volume to minus infinity or use the mute button. Muting will appear visually with no room for mistakes. That being said, complete silence is not very natural sounding and sometimes muting by lowering the level near minus infinity, but not complete silence will be preferable, changing and track spanning can be used creatively as in effect, and can also be used to refine the place of a track throughout the song. As an example, if there's a verse with an acoustic guitar panned, right, accommodating a vocal. But in the bridge, the guitar is delete instrument. I can center it and make it more frontal. Another thing that's important to note about panning is that sometimes because of the phase relationships between channels are tracking feel as if it sits better in the mix in a certain spot in the panorama. So it's worth sweeping the pan pots around and listening to where the track fits best. Automation has an artistic side to it and a technical side to it as well. I find myself using the cursor to automate delicate volume moves like vocals, siblings, and fine-tuning words to stand out in the mix. On the other hand, I find myself writing automation with the artistic moves more than the technical ones, like writing sense the reverb or delays, and just modulating parameters inside an effect because it's just more fun and musical when you perform it and do it like a musician playing an instrument when you want to ride faders, you need to enable touch automation on your truck. You can also combine the two approaches by writing faders and then doing the more delicate and accurate work with a cursor after the fact. Effects Sends, panning or plugin automation will usually be a part of the tracks sound and character. Therefore, it can be regarded like any other processor and it can be introduced in early stages of the volume automation, on the other hand, should be the last thing you do in the mix because volume automation fixes the fader and doesn't let you move it freely until you reach the last stages of the mix. This flexibility is somewhat necessary. So I suggest following this guideline. If you don't know how to automate in your DAW, figure out how to see automation parameters and how to enable touch automation in order to do this segments tasks. 28. Automation - Assignment: These desks might seem small, end-all, but doing these things in your mixes will make them more accurate and give them life and will serve you arrangements as well. Start by automating the acoustic guitars volume with your cursor. Trucks can sometimes need volume automation even if they're compressed. And this will be a good example for a case like this. But note to yourself that we don't make sacrifice over audio quality just to save us time. So go down to small details and make sure it sounds great. Next, open a reverb effects drug and a delay effects track, set them in a musical way, automate how much and when guitarists and to them, you can start by sending one beat in a bar with a cursor and then ride the sand while you listen to the track, try making this musical and beautiful, then automate a plugin parameter. It can be reverb or delay times EQ parameters and even compression parameters. This is easier on certain DAWs and harder on other. But whichever software you chose for yourself, knowing this will make your mixing possibilities practically endless. So learn how to do this before you move on to the last task, automate, mute and pan parameters of any track, any way you want just to be familiar with this possibility. 29. Mixing Techniques pt.1: In the upcoming segments, I want to introduce a few mixing tools and techniques you'll find very useful, again, most out of these segments, if you will stop after each technique that's presented and invest a few minutes just experimenting with it. Out of all the techniques that I will be presenting only to require third party plugins that you may or may not own. Even if you don't own these plugins, it's still very beneficial to know them and understand how they serve the mixing process. As the name suggests, parallel compression is the process of compressing attract parallel to the original uncompressed track. You can either duplicate your track and have one compressed or create an auxiliary track with a compressor inserted on it. In this case, you'll need to send your track to it pre fader so the level of variations won't affect the compression. Some compressors have a mix knob that will blend the uncompressed signal with the compressed one, saving you with the hassle. Just like normal compression, parallel compression can be used either to accentuate dynamics or to limit them. Big difference of having the original track as a part of the total mixture. If you set your parallel compressor with a slow attack, you will receive a transient enhancer that you can then blend to taste. If you've set it to a fast attack, the transients will be crushed and the channel will just become an overall level enhancer since the level you will place it in will be the lowest level that the track will reach. Here's an example of the two approaches. Parallel compressor can also be used to compress several channels and even the whole mix just to raise the overall level and to glue the mixed together, so to say. And additional useful parallel processing is distortion. You can set up a distortion unit and send audio to it just like you would do any other effects track in order to accelerate tracks and enhance their harmonic content. An important thing to note when you use parallel distortion is that because of the change in the harmonic content, the waveform changes as well and phasing might occur. So make sure to check your phase relationships post-processing, I usually have one EQ before the distortion, so I can carve the audio coming into the distortion unit and another one after the flip phase if needed, and to further process when necessary. Such in compression is the process of triggering a compressor on one track by bringing audio from a different track. Bass drums are very commonly used to such incense and harmonic elements in the playback. For example. This technique is very popular in EDM and pop tracks, but it can be used in acoustic mixes to create more space between instruments with overlapping frequencies or as a creative tool, you can create room for the vocals by placing a compressor on the playback, having it slightly compressed when the vocals come in. You can also compress a bass track when the kick hits in order to keep your low frequencies and check, each compressor will have its own way of setting the side chain to an external output. And once you have got that sorted, you will just need to send the source to trigger it. The term telephone filter refers to an aggressive cut of both high and low frequencies on a source that results in a signal that sounds as if it's coming through a telephone. There are no hard and fast rules to where exactly the filters need to be placed. And you can further tailor your telephone filter with peaks and notches or simply carve it to fit your sources needs. I swear I didn't speak as the mike be spin tonight. I swear I didn't speak as the mike be spin tonight, this processing can stay static throughout the whole mix or be automated in and out to emphasize ports or create dynamics. Sometimes you want your reverb to sound long, but not to actually have the whole length of the tail to prevent it from getting your mix all muddy. Putting a noise gate after your reverb will give you the privilege of having the initial decay of a long tale, but cutting the single one, it's no longer necessary. 30. Mixing Techniques pt.2: If your mix has a very distinct reverb to it, sending a bit of the delay signal to it might help get it closer to the ambience you're working toward. You can also try sending reverbs to delays or any other effects that you have set up and see what happens. Just make sure that you don't send a channel to itself by accident. Otherwise, you'll end up with a feedback loop, triggers or drum replacement plugins that analyze incoming audio and trigger a chosen sample from there available library. You can use them to fully replace a drum track or to work in parallel with the original channel as tonal support, MS stands for mid side, as you might recall from the recording chapter. And MCQs will indeed give you the option to process the middle and side's signals of a track separately. Midside processing is often used very lightly since these EQ create pretty strong phase shifts and affect the model compatibility of your mix. Dc queues can be used in a mixed or stereo tracks on group tracks or even the whole mix to cleanup signals and enhance width. Some DAWs might have this feature in their stock plug-ins, but most want. So you might need a third party plug-in if you have guitar pedals or an external effects unit, and your interface has more than two outputs. You can actually use them in your mix. Create a send that is routing to a free output jack. Connect that output jack to the unit and then record your processing to a new track. If you're using guitar pedals, you might need a ramp unit in order to convert the signal back to low Z. But even if you don't own one, you should still try it out and see if the sound you reach or satisfactory. 31. Pre Mixing pt.1: This segment will be demonstrated on Pro Tools, but the practicalities don't matter as much as the approach. So hang around. Regardless of your DAW, we will talk about importing, editing time of audio tracks and then get into fades, cross fades, compiling audio, editing workflow tips and mixing templates. If you are mixing a track you did not produce, makes sure you know the audio sample rate and open a project accordingly. If you import audio from a different sample rate to your DAW, it will either do a conversion which you can save the audio from or play the audio back at the wrong speed. If you personally produce the track, make sure you save as and name it. Something like blah, blah, edits lets you edit, knowing you can always go back by simply opening the clean session from the pre-production phase, editing is just as important as recording and mixing. So don't rush through it. The groove or feel of a song has a big impact on the way the listener perceives it. If you want the song to move the listener in a certain way, you can create that throughout the editing stage, there are a few editing approaches that will change by the aesthetics of the song you're working on. Pop productions will usually be edited 100 per cent of the grid. Whereas in the or R&B songs have more fluctuations between the beats, the groove of the song will be lost. If you just snap everything through the grid, it's important to work with your ears and now with your eyes when you add it in order to get things flowing musically, rather than precisely editing rhythm can either be done by cutting and moving the audio or by warping cutting the audio will leave the waveform as it is, but change the audio track. Warping, on the other hand, will change the waveform but leave the track intact. The two techniques have their positives and negatives and therefore are used under different circumstances. Cutting and moving audio will create either a gap or an overlap of audio that will need a crossfade. Warping will not require cross fades, but manipulation of the waveform itself might introduce audible artifacts. Although the algorithms for audio warping are always improving. As a rule of thumb, I tried to keep my waveforms as they are at edit by cutting, moving and cross fading, there might be a bit more work, but it will sound better and in my opinion, it is worth the effort. I do use warping when I edit an instrument with harmonic content that can be cross fitted without it being noticeable. This usually is the case with instruments that sustain long notes, but this rarely happens. And I estimate that 99% of the time I edit audio by cutting. Another scenario where I will use warping is if I need to import a loop that is not distinct to my projects tempo, if this happens, it will be during the pre-production phase, as mentioned in the beat making segment, when you're editing, you may want to try working with your grid turned off. This will make sure you really listen to the music. And here, if anything, is a bit off rather than analyzing it visually, that being said, the grid is very useful and can be used to create clear reference points. An important thing to note is that when you are editing an instrument that is recorded with multiple mikes, you should group the tracks together and edit as if it's a single track, fades or quick volume changes, a fade in. We'll take the audio from silence to the track's volume. Likewise, a fade out. We'll take the tracks current volume to silence. Fades can be tailored to many lengths and shapes, giving you the ability to make them sound as natural or unnatural as you like. The reason fades are important is because when audio is cut in the middle of a cycle, a digital click or pop occurs. Fading the audio in or out. Solved this problem, cross fades are used when you want to create a transition between two audio clips. The first clip is being fitted out while the second is faded in, creating a smooth transition when worked correctly, since fades change the track's volume, it is important to note you should not be fitting any transients or other pieces of audio that are important to leave as they are. So even if you use your DAW's automatic fading function, go over and make sure that you didn't fade something that should be left as is compiling or in short, comping is combining audio from a few takes into one performance. This is a very useful process that gives you the option to take the best pieces from each of your ticks when you're comping any instrument, you should think about the small things that are a part of its character and make sure they're not left out. A good example of this is when you're comping vocals, you need to notice that you add the breaths to your compiled track. You should also notice not to leave out essays and so on. That being said, during the copying process, you can start cleaning your tracks and leave out mouth noises or excess pieces of audio that have unwanted noise in them. If there's a need to pitch correct vocals or any other instrument, do it before you mix, whether it's manual correction or an auto tune plug-in needing to stop mixing to make these corrections might break your concentration and kill your vibe. Tracks might have unwanted noise recorded to them, either by faulty piece of gear or electricity issues that may have occurred when the music was tracked. If that is the case, you should trim the tracks to the point where only relevant audio can be heard. This is also useful when the noise is faint because if you do have a processing on a track, it will become very apparent. So it's better to deal with these issues before you start mixing phase relationships and multimedia instruments can make a big difference in the recording stage, we're doing as much as we can to ensure phase correlation. But through the computer after the fact, we can greatly improve the accuracy and therefore get better sources to work with. If you did not record the tracks, you should make sure that multi-track instruments have healthy phase relationships that don't harm the low end or timber drums will be the clearest example, since there are many mikes at varying distances, I like keeping my bass drums phase positive so the audio pushes the speaker outwards when it hits after, I make sure that's the case in the overheads and bass drum close Mike. I use that as my reference point and go track by track, making sure that this initial phase correlation works. Some people stopped there, which is fair. But if you want, you can go a bit deeper. These are the bass drum, snare, and overhead channels of a drum set. You can see the phase in this bass drum hit is positive in all tracks, so they are in phase. But if you zoom in, you can start to see that the close my channel appears a bit earlier than in the overheads. This makes sense since the close mic is physically closer to the bass drum and therefore receives the sound waves faster. There are plugins that will automatically do the compensation and timing differences. But if you don't have access to these plugins, you can simply drag the audio back and align the two phases. The differences can be anything from subtle to not subtle at all. But even these slight differences make the audio work better together and save you from doing unnecessary processing. It is important for me to note though, that there is no such thing as a right thing to do in the audio world. So if you like the sound of the less accurate or even out-of-phase version, go for it. The only thing I do recommend is making an effort to listen to them both before settling on one or the other. This is relevant when you send tracks to a mixing engineer, it's important to render your tracks from the same starting point as it's doubtful that they will be able to guess where a certain piece of audio is supposed to be. So even if there is a lot of silence and attract, you'll need to render it and export the files knowing the mixing engineer has no room for error regarding where things are supposed to be happening if you are mixing on your own but in a different software, this process is still needed. However, this step can be skipped if you'll be mixing in the same DAW, mixing is a very intuitive process. So having control over the session is extremely important. Being well-organized will make your job much easier and more enjoyable, and therefore, probably better if you are mixing a song that you did not produce, make sure the tracks are in an order you are familiar with and labeled simply and intuitively so we can find tracks easily when you look for them. I also like having markers and loop selections in the project that let me know where I am in the song and allow me to find a specific part quickly. These are things I make sure that are there and clear before I start mixing. Lastly, if you have a mixing template, this is also the time to import it and make sure everything is routed properly. We will talk about mixing templates in greater depth. Further in this chapter. 32. Pre Mixing pt.2: Here are a few tips to make your editing process better and faster. Make sure you take the time to find where the features I mentioned are in your DAW and what they're shortcuts are playing with the waveforms size will help you notice the fine detail of the audio track, like breaths, noises, and other quiet information that will be hard to hear when played with an unaltered waveform. Zooming in and out will give you the ability to do more accurate edits, since you can really pinpoint where you are affecting the audio. If you want to get comfortable with really fine details, you will need a full view of the audio you are working on. Besides zooming in, you can enlarge the track you are working on. So it takes more space on your screen since getting audio is likely to be a big part of your editing, being able to access and use this tool effectively and efficiently is essential while editing many audio files, it's almost certain you will be using a lot of fades, which will take ages to do manually. Using this shortcut will allow you to work a lot faster if you cut a piece of audio and want it snapped to the closest bit on the grid. You can use this shortcut instead of dragging files with your mouse, you can decide how many beats the grid displays in a bar. This is useful if you're working on 16th note beats or any other measure. But sometimes you also want less visual information on your screen so you can quickly change your grid resolution when you're editing audio, you'll sometimes need to be between beats, figuring out how to jump quickly, between automatically snapping to the grid and having your cursor free will help you do that. 33. Pre Mixing - Assignment: The assignment for today is to take a project, edit it, and prepare it for mixing in the R&B folder, there is a folder named unedited drums, import the whole song with the drums and start prepping. I would start with phase correlation and then edit the time and groove if you feel it's needed, make sure you fade or crossfade every edit you do, and make sure that fades. Don't go over transients. Lastly, try using the tips mentioned in this segment to enhance your work. 34. Monitoring pt.1: You're monitoring levels or in other words, how loud your speakers play have two important impacts. The first is related to a phenomenon which was first described by the Fletcher Munson curves and now is described more precisely by the equal loudness contours. These graphs show the humans ear's sensitivity to different frequencies in different levels. As you can see, our ears are far from linear and as the levels change, the frequency perception changes as well. We are very sensitive to the mid-range frequencies and our sensitivity to low frequencies increase as the levels get higher. This raises a very important question regarding our work, because if we tailor a mix, knowing it will sound different. If it's played in different volumes. In which volume should we mix? My answer is mixing all volumes, but let's look at it in greater depth. You should get used to a specific initial starting point for your volume knob and get familiar with it. This starting point is suppose to give you a comfortable SPL volume when you start raising faders. That will also translate with healthy peak levels down the road. For example, with my setup, I know that when I set the volume knob at this position and raise the kick drum fader to hit around minus ten dBFS. It will feel comfortable in the room and it will be around 85 dB SPL. As channels are added, the levels will rise. Ten dB is enough headroom for you to mix without worrying about clipping, make sure that you set the initial levels while listening to the loudest part of the song. So you won't end up being surprised by level's changing from the arrangement after this first balance is put together, you can jump between listening to the mix loudly, listening to it quietly, and making sure that it works in any scenario, it will be different and that's okay, but make sure that it's not inferior when you want to fine tune the low frequencies, raise the volume to around 85 dB SPL, because as we've seen in the graph, that's where they are the flattest. The second important thing to note regarding monitoring levels is ear fatigue. Our ears and brains constantly work and are in fact wearing out during the day, especially when you listen to in high volumes. Here, fatigue will lead to an alteration in how we perceive high frequencies and dynamics, which will in turn make us do wrong. Mixing moves, your ears, stamina will increase over time, but even well-experienced engineer's take breaks during mixing days and stop working if they notice their ears are tired. So note to yourself that working on a mix ten hours a day does not make you a devoted professional, but actually harms your work because from a certain point onwards, you will only harm your mix. Stopping for a few minutes every 45 minutes or an hour will help with both ear fatigue and regaining perspective on what you've already achieved. You should also finish your working day. If drafts you thought were sounding great, suddenly appeared dull and flat. The higher the volume you monitor through, the faster your ears will get tired. The fact that we can have one speaker playing different elements from the other speaker might give us a bit of an easy way to avoid, difficult to work. Although panning and panorama are great artistic tools, summing down your mix to mono can be beneficial because when both speakers play the whole session, you can really hear whether your mix is spacious or cluttered. Another reason listening in mono is beneficial is that your mix can be played in mono from a phone or mobile speaker. If you want your mix to translate well in these circumstances, you need to make sure it does. Mixing studios usually have a few sets of monitors because every monitor has its own properties and character and referencing mixes will make sure that it translates well in a variety of configurations. You can work on inexpensive full bandwidth monitor and then jumped to a cheaper monitor that lacks both highs and lows. To realize that something in the midrange is not exactly right yet, these small mid-range monitors are often regarded as a **** box and are used in professional studios around the world. Additional monitors are very useful when you mix, but they are luxury that not everyone can afford. Balancing your track and listening to them through your phone, laptop, speaker, or any lo-fi setup will also be very helpful and should be done even if you do have a few monitors in your studio. There are also plug-ins like the mixed checker that gives you a few algorithms of different cheap speakers. You can check your mixin. 35. Monitoring pt.2: When audio is played in an untreated room, frequencies can build up or cancel each other out. This essentially means that the audio that ends up reaching your ears is not the actual audio coming out of the speakers. And since you can't mix what you can hear, monitoring in a way is the most important thing to take care of. Here are a few pieces of information that are very important to know and that will help you setting up your own mixing studio. And mixing happens. The term sweet spot regards where your ears will be when your work and it's supposed to be the flattest spot recording audio alteration coming from the room itself where you place your monitors in the room and how you position yourself in front of them will have a greater effect on what you hear. So here are a few guidelines for setting up your sweet spot. Your monitors should be set in a way that will create an equilateral triangle with your listening position. Or in other words, that the distance between you and each monitor should be the same as the distance between the monitors themselves. Try placing your monitors between two walls that are built from the same material and in front of the wall that is built consistently as well. The next guideline will be placing your monitors at different distances from the back wall, the side walls, and the ceilings to prevent frequencies from extreme picking, it will also be helpful to have the height of the speakers above or below the rooms center. Every room will have it's own resonant frequencies due to its shape, size and the material it's built from when you want to find an initial place to start your rooms. Acoustic treatment a suggests placing your speakers on a stand and playing song you feel you know well enough to notice audio alterations and make sure your ears at the Twitters height or between the Twitter and the Warfarin is height. Sit centered between the two walls you chose for your sweet spot and create the equilateral triangle with the speakers. At this point, play the song at a moderate level and try noticing how pleasing or unfreezing the current distance is. Then move the speakers backwards or forwards in the room, repeating the process until you find a place that you feel is neutral, that will be your sweet spot. If you're in a small room, note that you should keep away from the rooms center, since that will be the most problematic regarding frequency clashes. Professional mixing studios have acoustic treatment that makes sure the room does not affect the audio coming out of the speaker and into our ears. If you have the privilege and access to a treated facility, I highly recommend mixing there and not at home. But if you're starting out and still need to train before you charge people enough to afford these studios, make sure you acoustically treat your room as soon as possible. Every room will require different treatment. So it's practically impossible to give a hard and fast rule that will apply to all spaces. But here are things that you should know and can help you with the basic acoustic treatment of your room and hopefully get you closer to the point in your career where you can earn enough to hire a professional to do the job. Acoustic treatment will consist of absorption and diffusion. Absorbing materials will either be acoustic foams or acoustic panels filled with dense rock wool or fiberglass thin phones will help absorb mid to high frequencies, but won't be of much help with low mids and bass frequencies. The purpose of diffusion is to scatter sound waves and spread their energy instead of reflecting them back and have them echoing in the room. This results with a more pleasant reflection and a better standing room with less peaks and resonances, bouncing and altering your listening sweet spot. Next important piece of information is that base tends to build up in corners. This is why many bass traps usually break the corners shape and mixing rooms are rarely square. Base frequencies aren't as directional as high frequencies are, meaning high frequencies won't mainly propagate towards the direction in which they are projected to. This is why the sound gets duller when you stand up in front of your speaker, instead of sitting down, if you stand by the side of your speaker, it will even get duller. And if you go behind the speaker, you will mainly hear low frequencies. These base frequencies bouncing from the back of your speaker are bouncing back from the wall to the room and your ears. So addressing them will also be an important element of treating your room. For starters, start with treating the first reflections of the room, which are the walls from the right and left of your speaker, as well as the rear wall and the ceiling above the speakers. These first eight moves will make the biggest difference and are definitely a great place to start. If you hear there's a need and you can afford them by bass traps and place them in the corners for the sidewalls and the ceiling, you can use foam, but as we mentioned, they won't be as useful with base frequency. You can either buy an acoustic panel or build one yourself using any DIY tutorial on YouTube. It can be fun and give your home studio a very personal touch. With this minimal treatment, I would generally suggest using panels more than foam because they are generally more effective. But either way, make sure that you placed the absorbing material at a height that will place the speaker in their middle, that will enter result with more absorption. My first home studio setup consisted of two DIY panels on the right and left of the speakers. One heavy panel behind the speakers that could handle low frequencies, two triangular base traps in the two corners behind the speakers, and a few pieces of foam that were glued to a wooden plate and hung from the ceiling with a tilt. Lastly, I placed to diffusers behind me and the same height as my head, hoping that the frequency is coming from the back wall will be projected better through the room. This was not an ideal mixing studio, but it definitely lead to better monitoring and certainly great for the time and place I was in my career. 36. Template: After learning how the tools work, building a workflow and some logic behind your mixing approach will be the next big step. I have personally invested a lot of time in researching and experimenting with workflow and mixing approaches. And we'll now present techniques and concepts that you can experiment with and see what works best for you. I'm mixing template is a saved format that you've tailored to your needs that can include routing, plugin settings, and visual settings for your sessions. This can be imported to every new mix you are working on and give you a good, solid foundation to begin with your initial preferences already in place. A basic template can include buses or group tracks for the songs, musical sections with Sends ready for preset effects like long and short reverbs and delays all pre routed to the master bus when these few ox channels are important to session, route the audio to them and then you are good to go. But this can even go further. For example, if I know I tend to use triggers when I mix, I can have the trigger plug-in ready as a part of my template for both the bass drum and snare, I find myself using parallel processing like compression or distortion quite a lot. I can set that as a part of my template as well. And that will save me from opening and routing these channels every time I mix. These adaptations are endless and can be shaped and tailored to what your DAW has to offer. My mixing template includes triggers, separate parallel processing for the bass drum and snare parallel processing for both, and additional parallel processing to the whole drum section when I feel it's needed, then each section has its own long and short reverb, long and short delays, and another sent to a Weidner plugin. Then each section is routed to its own stem Bus, which makes exploiting stems very easy and fast. These can all be sent to a parallel compressor and reverb, which are always there, but I use them only when they are necessary. All the stems and the parallel processing are then routed to a master bus that is also ready for processing if required. I have built, changed and refined this over the course of a few years. And I'm still constantly revising whenever an idea or change in my workflow comes up, as I've pointed out a few times, speed keeps the creative juices flowing. So working on a template of your own is totally worth the time and invest it. You can take any idea you like and experiment with it to see how it fits your mixing workflow. 37. Mixing Workflow: Workflow and mixing regards both in approach and how you divide the mixing tasks. One of the first things to think about is how you start your mix in order to finish it with healthy levels without flipping your master channel. Some engineers like setting a reference point, like I mentioned, the bass drum and have it hit around minus ten dBFS, for example, knowing that even after all the other instruments will come in, the levels will still end up below 0. Try finishing mixes with peak levels before minus three dBFS. So there will be enough headroom for further manipulation of the audio in the mastering process if needed. If what I said is not totally clear, go back to the level and metering type segment in the recording theory chapter. The first monitoring segment in the mixing jester. Besides thinking about your initial levels, you'll need to think about which channels you start your mixed with. There are a few approaches when it comes to this subject as well. You might have heard about top-down mixing, which is an approach that suggests starting your mix with processing the master or group buses and getting your mixed dotted with broader brushstrokes, so to say, and a macro perspective. The other approach we will be starting to work track by track, maybe start from the songs. Leak channel may be the vocal or any other instrument and only then add channel by channel and piece of the puzzle together. That way, many mixers actually start by mixing the rhythm section, building the song's rhythmic foundation, and then move on to either the vocals or any other channel or section they feel is relevant. Each approach has its pros and cons. Mixing top-down is faster than mixing bottom-up. Bottom-up mixing will give you more familiarity with the sessions tracks and maybe lead to more precise adjustments than the broad processing involved with the top-down approach. That again, hearing everything in context is very important since the tracks interact both musically and with their phase relationships. So top-down might give you a better overall perspective. As you see, there is no right or wrong. And I can only share these pros and cons and in approach that might be useful in taking the best of both. So here's a practice I found very useful in the past and still find myself using from time-to-time. Although my current mixing workflow is a bit more flexible, the practice is separating the mixing process into three stages, corrective, top-down and bottom-up, starting with all the faders down, the initial corrective stage starts with a bottom-up manner. You raise faders one-by-one in order to create an initial balance. But you approach it with the mentality of a recording engineer, not yet a mixing engineer. You go over the tracks, noticing how channels interact with their phase and tone, trying to understand which channel service, what purpose, and trying different balance variations. Insert plugins unless you're really sure if they are needed and if you do insert something due. So thinking you're creating a source rather than correcting one. In this stage, I also set my effects tailoring, reverbs, delays and modulation sends to each section of the song. Then, after I'm done creating these sources and effect palette, I already have a good balance and am very familiar with the sessions tracks if you haven't done so until this point, this is a very good time to take a break and give your ears some rest when you return, giving you a reference track, Alyson, this will recalibrate your ears and after giving your current mix and other listen, you'll be more confident in setting bus processing if you feel it's necessary. This starts the second stage of top, bottom. It's important to note that processing a group or a mix is different than mixing single tracks, both technically and artistically. A simple example will be that processing a group will require a lighter, more delicate approach since 0.5 dB changes appear in all the tracks you process and not only one in this stage, you can also make the decision to set parallel processing on the whole mix if you find yourself liking it, this concludes the top-down stage. And so we move on to the last stage, which starts with a very surprising move, dragging all the audio faders down back to minus infinity. Doing this after all the effort involved in the first two stages might sound counter-intuitive, but I can tell you that this was an amazing addition to my mixing workflow. And I sometimes repeated in a mix more than once. The reason is that it's like starting from scratch regarding your perspective, but with amazing sources, custom tailored effects, and custom tailored bus processing for your session, which is an amazing base to start a mixed from. If you ask me, as you rebalance the tracks, try bypassing and I'm bypassing what you've already done in the first stage just to make sure that these adjustments are actually helping. You may feel the need to further tailor them or decide they're not necessary at all. And after you're done with the additional processing, take another break and play the song from the beginning, set the balance of the first part exactly how you want it and then play the whole trap reading volume automation, Effects Sends automation is something that I do during the mix because it has a lot to do with the tone and emotional movement that the song will have. But with that said, in the last stage of volume automation, I do find Effects, Sends, and returns. Keeping in mind that after mastering, they will get a bit louder. After I feel as if the mix is done, I take another break and I come back and give the song and listen. This is a very interesting position to be in, since you have decided it's done and you're ready to print it out, but you still have the option to change it if you decide to do so. My suggestion is, don't change anything. Your ears are already fatigued and your perspective is long gone. Come back to it tomorrow and give it a listen when you're fresh. This sums up the few workflow approaches you can and should try out, experiment and figure out what's comfortable for you and what gives you the best results if you managed to combine the two, stick to it. 38. Saving, Revising: There are a few things to keep in mind when you've saved export and revised or mixes that will keep you on top of your work and might save you a lot of time. You should create a naming system for yourself to track your work. For example, I name my first mix, blah, blah, mix v1, v standing for version. And when I revise the mix, the first thing I do is to name it V1, 0.1, meaning it's the first Mix, first revision. Then if I decide to start the mix from scratch, I name it V tool and start the revisions over, blah, blah, makes V2 0.1, etcetera. Having a naming scheme and making sure I save As every time is important for two reasons. First is you don't really want to keep working on the same project over and over because you might figure out that one of the early mixes was the best one and you won't be able to recreate it. Having sessions saved separately lets you see your progression and sometimes you can import specific elements from different versions and use the best of a few. The second reason is that you want to know how many times you have worked on a project for the clients and your records revisions are an important part of the mixing stage, but your pricing scheme might include a specific number of free revisions. So having this documentation gives you the indication of how many revisions were done and gauge pricing accordingly. Mixing my gate, you focused on such small details that it's almost impossible to nail 100 per cent of the work in one go. Revisions are an important part of mixing, but they can also be a dangerous process. There's a difference between a poor mix that needs another goal and a good mix that just needs revision. A work of art is never really finished. And so you need to approach the revisions knowing that your goal is to finish the song that you feel is almost complete. If you end, the artist agree that the mix is good and just needs a few tweaks. I suggest listening to the song together from the top and making a list of the things that you feel need a dressing as you listen through. If you spend a whole day revising, you have either ruined good mix or maybe you're good mix, so it's not so good. After all, luckily, you have saved the project under a different name so you can go back and decide which is the case. Before you export your mix, you can put a limiter on the master bus and push the levels of the mix a bit higher. Because when we mix the levels are below 0 dBFS. We need to gain it up so the client won't compare your mixed to a finished mastered track and be frustrated with the fact that the levels are lower, but don't slam the limiter too hard because it will alter your dynamic range. And you do want to keep the mix and tax four to five dB of gain reduction should be the maximum you go through, add your name to the exports and label them in a way that will correlate with how you label your sessions. This will save you unnecessary confusion. And finally, not the processing that you have done for it. For example, blah, blah, no man mix, V1, 0.3 limiter. The correlation lets you easily follow your own trails if you need to. And having your name embedded in the file is both good PR and a way to prove your affiliation with the track. If sadly, you end up needing to prove anything to anyone when you export the file, make sure you export at the same sample rate. You mixed in as well, ensure you export an 24-bit and as a stereo file and not dual mono, you can export the limited file in both MP3 and wave. Sometimes artists need the mix, export it as stems, which are separate exports for each section, stems are often used for playbacks in live shows to create remixed versions or for stem mastering needs, which is a mastering process that works with a song divided two stems. Instead of working with one stereo file, artists might have different requests regarding how they want their stems divided. Maybe they will need the bass and snare drums separate from the whole drum. Stan, maybe separate effects stands for the vocals. I planned my template in a way that will be easy to stem out and I make further adaptations when they're needed. You can decide for yourself if printing stems is a part of your mixing service, which will give an additional over-delivering factor to your name. But it's a process that might take some time and additional work. So it's commented charge extra for stem exporting. If you send your mix to a mastering engineer, you should export in another file with no limiter and the projects properties noted in the file's name. For example, blah, blah, nomad mix V1, 0.32444.1. This will let them know how they should open their project in order to maintain the correlation with your naming scheme, you can then send both the unlimited wav file and limited one as a reference, if you like, how to eliminate affected your mix. This concludes the mixing chapter. So for the last segment, I want to go over mixes. I've done and talk you through both the technical aspects and the mindset involved in them. Combining all the theory we talked about with practice, both regarding the technical and mental approaches that were brought up is the most important thing to go over. So let's get to it. 39. BirdSchool (Get Familiar With The Song): Holy and find the yellow around us like Kevin, when you forgiven of questions. Holy. Molly and my soda, floating color and Baby Yoda. Yoga, meditation, masturbation and bread baking reality has some shape. Problematic systematic confusion live in the reality in your body. And trigger point that the kidney cannot. Mission versus deficiency, a tiny, efficient as deep, a poisonous car. What is really real road? Is it telling me what did the selection of emotion will drown me? Reach this guy? So someone said, flowers bloom where they should. Even when concrete is dry, when everything is done. You could pick up and down and never leave you to read it by gravity with formalities and courtesies and how much afraid or you, how much prepared. This very thing you'll carry on carrion. You know, they got a plan reasonably guy that someone's home or they show even when concrete is dry. Everything is everything. Everything is the event horizon being, as it were, lost in a dusty hay as ****? Humanity is left stranded in present tense. Bereft of a visionary future. An island in a sea of timeless same as the profits knowing Divination Records, electrolytes, procrastination. Tradition plots the path. There isn't really any place to go. There never was. The means to the ending we are told to remember is indeed a ****** one. Our constellation, the urge to the verge of happening trending towards a nostalgic future. It would almost be tragic if it weren't so **** funny. 40. BirdSchool - Mix Overview: Now that we are familiar with the arrangement, we can dive into the mix, I will mention again that my goal is to expose you to tools and techniques. So you might see plugins that you are not familiar with. The names of the plug-ins are mentioned in the segment and you can look them up if any of them catch your attention. There are a few approaches when it comes to mixing drums. The first is to use the overhead and roommates for the overall tone of the drums and use the close mikes just for transient and mild tone shaping. And the second approach is to use the close mikes for the dominant tone shaping of the drones and use the overhead and rooms for just tumbled users. Because these germs were recorded for a different song that require different Sonics. Imported these drums just for their part, but not for their sound, required quite a lot to bring them where I wanted them to be, and a few very non conservative methods. The main track I used as an overhead is actually the rooms I didn't use. The overhead jacket just didn't suit this production. And it originally sounds like this. And now with processing, pretty extreme, I know, let's go through this step-by-step. First thing I've done was to tame the highest here. And apparently there was a resonance that bothered me. I'm assuming I have inserted this after, I've inserted this distortion. You can notice that the distortion brings a lot of aggression and kinda compresses it. I'm using this more as a compressor. Let's bypass the pre distortion EQ, and on bypass it while the distortion is on to notice what the effect is. The higher kind of pops out and it's unbalanced. And more than anything, I think this suits the R&B kind of textures more than anything. Then there is an instance of trash, which is another beautiful distortion and filtering the source before it goes to the saturate or which I see is pretty violent and then compressing it. Alright, so let's give this a listen. The first AQ probably came in after I've inserted trash because the high frequencies are wild and it really brings up aggression and squeezes completely different emotions from this track. Then I've inserted a course, which is not a very common thing to do on an overhead. But let's see what this does. As you might have noticed, the mix knob is dial down to 14 per cent. And what I feel it does is it creates more dimensionality and more depth. It's not really noticeable, but it's still does a very good job after the course, I'm putting a phasor. This is thirty-two percent wet. Let's listen to this. This is a bit more noticeable and it does create this movement in the frequency spectrum which we can feel and hear. And finally, this dq. So I'm filtering these frequencies from the side information, which means that all my low energy, we'll be in mono. Then I'm also filtering from the whole panorama, 90 hertz and below, and taming the high frequencies. Again, let's listen to this. Kind of plays around with again, the aesthetics which suit the lo-fi R&B field and where the kid is placed inside the room feels different. Now, notice this feels as if he's closer to us and the room is a bit smaller. And I'm sending into a Weidner the studio D to create more dimensionality. Let's listen to this. This is bypass. This widens the sound is actually another course. I'm not using this track a whole lot in the whole mix. As you can see, it's minus. 38.5 dB in the verses and the choruses. I think I automate it a bit higher because there are more symbols. But in this mix, I've used the close mikes and triggers as the main elements of the drum kit. So let's go to the kick. I've actually not used the recorded kick at all. And the reason is, as I mentioned, this was recorded for a different song that required a different feel and different tones. So I've used drum replacer. This is my current favorite trigger plug-in. I've inserted two triggers samples. This is the first one. And this is the second. Both of them together. Now let's listen to how this interacts with the wrong track. They are doing the heavy lifting and simultaneously, there's a parallel chain of the bass drum. You can see these two are routed to these two auxiliary channels and it's being distorted and then filtering it out. So let's listen with and without this parallel chain gives it just a bit more weight. We're according to snares. There's this and this and oh my god, I've been through a lot with these narrow channels. The original snare is this. This is distorted in every stage possible. We actually recorded this specific track through such a vintage preempt. It did not have a volume knob or fader. I don't think I would have done it if I would have tracked it today. But anyway, this has a lot of hi-hat bleed. The hi-hat bleed was really, really accentuated by this processing and it had a lot of impact over the whole groove. So starting off, I've cut all the sub frequencies which were booming for no good reason, nine compressed using this beautiful Compressor, Let's do an AB test. So you might have noticed that the high hat kinda comes closer to the snares peak. Besides that, there are two snare triggers. The first one is this, and the second is this. These two samples are way lower than the original snare. These four channels are all routed to this one snare bus. The reason is eventually after I do this balance between these single channels, I just want one fader I can regard as my snare channel. What I've done is firstly, insert this beautiful as a cell plugin. So I'm raising three dB at eight K, but with a bell, the classic SSL move is to raise eight k with the shelf I'm using the belly was more punchy. Next off a transient designer, I wanted more attack. Listening to it again, I don't really love it, but it's really important to note that signal chains are built together. We might understand what this is doing only by the time we finished the whole processing chain. Then there's this instance of Saturn, which is a multi-band saturated. This is a warm type. These are all old tapes and this is also o tape. I see, the main thing I've saturated is this low mid band. Let's see what this is doing. Cool, gives it a lot of weight. And the beloved sketch cassette to kind of feels as if it's taking the pitch of the snare down, it's giving an even more weight. You can see I'm degrading the quality of the cassette tape a bit and slowly saturating the signal by pushing in to the circuitry that makes nouns are dialed down a bit. Then another instance of trash, there are some filtering going on. Again, attenuating frequencies, pushing in high frequencies. In the saturation module, I'm using tape saturation, then convolve ever so gently, 5.6 per cent wet. I want to listen to what this convolution is doing. This is giving some weight in the lower frequencies and then compressing. Let's listen to everything. The overall mix fader is dialed down about 50%. Next up, I'm attenuating, I guess this is the fundamental and some mid-range frequencies. Let's listen to this. Gives the snare kind of smack berserk distortion, which I like using sometimes as a compressor. So this kind of brings back the low punch. And as you can see until now, and we still have one more plug-in, but there's always this game of like two step forward, one step back towards the forward, one step back, and eventually it arrives where we want it. Lastly, there's this omega transformer, this models a new transformer. Then there are these two samples to support the snare once and awhile. Let's give this a listen. It's just a sample of a clap which I filtered the low frequencies from an assemble of broken glass, which I filtered a lot of the high frequencies and the low frequencies from. It's just adding another texture. Lastly, this is sent to a parallel distortion. I'm using the preserve distortion. There is pretty distortion EQ. You can see that I'm using dynamic EQ here that is pushing these low frequencies up whenever the snare hits and then post distortion in queue with filtering of the low frequencies from the sides because I want the snares, low frequencies really centered. And these are frequencies have nothing to do with snares. Let's listen to this inserted. It's really subtle, but it just feels a bit more mixed and it's sent to a delay. Let's listen to what this delay is doing. I just sent this amount, so it's really, really gentle. Just to give it some more dimensionality. Let's listen to this snare with them to, based on triggers. I'm sending the whole drum kit to this compressor and it adds both punch and weight. The parallel compression does quite a lot. Lastly, the floor tongue, which is also a trigger, I did not feel like the original floor serves the intention of the song. So this is the Trigger I've inserted. This is actually from Steven Slate trigger. The last cool thing I can share with you about the drums is the phis that I've actually never done before. There's this fuss which is automated. Let's look at what this force is doing in the C part. Very violent, very cool. Moving on to the base. Originally, it sounds like this. Not too exciting. Before I've done anything, I've inserted a gate which just cleaned the noise coming from the recording and an LA to a, which is a great compressor, but it also kinda saturates the sound. Let's give this a listen. Gives it some more character, and then I've split it to two auxiliary channels. The first one handles the very low frequencies I wanted the base to have. And the other deals with the high frequencies. Let's start with a lower one. So I really want the base controlled. I've inserted another LH, which does this, brings some grid. I really love what this is doing. And finally, a multiband compressor which reacts to certain notes more than others. And again helps me keep the base in place. Then with the high channel, I'm having a lot of fun. The first thing I'm doing is another gate because I'm going to be distorting the **** out of this and filtering the low frequencies. Then I'm pushing the hymen and filtering out all the very high frequencies. This is how it sounds. Then I've inserted the studio D, which takes the source from being mono to being stereo. Now that I have this channel stereo, I'm distorting the **** out of this with a very, very extreme setting. Then I'm inserting this EQ, which helps me shape where the energy is. So I think this range was kinda saved for the pads and the drums. So I filtered this and let's see what this is doing. Keeps the energy a bit higher than the sketch cassette, which I see I'm driving the input of. So it's going to add more saturation, Some more aggression, and then taming the highs. And just setting where I want the focus of the energy to be. So very much low fi. And then I'm sending it to another studio, DT have it even wider. And with the best channel, a lot of energy. So again, with no processing, Let's listen to this inside the track and the yellow around us like Kevin, 24-seven. Very wimpy compared to heal around us like Kevin when he presented question. A lot of fun. The acoustic guitars in the beginning originally sound like this. And after some processing sound like this. I'm using the decapitate or a setting which is text sketch cassette, which is a different form of tape, cassette and the reserves. So you can see there are three different saturated colors on these two channels. And then some AQ. This is pretty wild. Let's see what this is doing. Just kinda makes it more low. Phi takes the lows and highs out, enhances the midrange. This is automate. Oh my God, Wow, what the ****. This is a beautiful, Let's see where this, where this automation comes in. Right? So there's this break before the course. This is where this setting comes in. So we have this sound in the beginning. And in the break, it's this. The electric guitar has a pretty crappy sounding to begin with, so I needed to kick them up a notch. This is the picky kind of guitar. I've inserted sketch go set, the old one with the NR comp, which is a very violent compressor and it's very, very bright. And I've added some wow and flutter which are fluctuations in pitch, which are doing this tool. Very vibrant than some EQ. Just enhancing the pic ever so slightly. These guitars or I have here. Let's see how they sound before. Pretty wimpy. What I'm doing here is some EQ, not too much, some saturation console saturation. And then there's this compressor, supposedly a guitar pedal compressor. And sketch this out again with the compressor engaged. And after two compressions and some saturation, it goes from two. You can see there's massive amounts of compression going on. Now, these are the ambient stuff from the intro. There's really nothing too interesting. Slight distortion. You know, nothing much, but the interesting parts comes in this path. This path originally sounds like this. And after I've worked on it, gradually. So what I've done is filter the highs and lows, heaps of distortion and playing with the tone here, making it a bit bright. A lot of energy than another form of distortion. These two are warm tubes. This is warranted, but I'm not saturating this at all. Let's see what this is doing. Brings more energy in the low frequencies. And this notice how this sounds way closer when the high frequencies and low frequencies are attenuated and the midrange is enhanced. It feels really, really close. Then I'm sending it to some delay, some more just for you to hear. And the trick is very fast LA Times, but quite a lot of feedback. And I've invested some time in searching for the right style and some Weidner, the studio D, I love this plug-in. Notice what this is doing is adding a lot of level but also enhancing the stereo width. This is the only thing we have in the first, in the chorus, we have this pad. Which arrived like this. Great sounding pad. I just wanted a bit more gradually. So another catheter distorting the **** out of this. Adding a lot of grit than attenuating the low frequencies and a bump on the hymen. Then there's the search and coming from the piano. And I see it's reacting only to a very specific frequency range. Well, listen to how the two interact in a moment. Another EQ, which is automated, automated this to give this more life and a sketch cassette. This just degrades the quality it takes it from being very hi-fi to being very low phi. And now let's see how this compression from the piano helps. This mix just gives it more space. The piano has only one EQ. Let's see what this is doing. It's just a bit fluffy without it, and just has a bit more high frequencies with it. The Mellotron has also just simple EQ filtering the high frequencies because we want to leave them to the vocals and the drums and also searching from the bass drum. And now let's listen to how all the elements of the playback interact in the course. And before we go to the lead vocals, Let's talk about the solo sense that come into bridge before the second verse. So we have these two sounds. So the first synth, I'm just filtering the lows and highs. And it takes it from this to this. Kind of keeps this overtone and check, then I'm doing this which seems a bit contradictory, but it's a very different filter shape. It's as if I'm doing this. Let's see how the second filtering helps us. I like what it's doing. Then chorused, mixed. Now at 43% of eighties vibes, this tape Plugin kinda makes things sound better when it's pushed as far as it can go. And lastly, some attenuation of the high frequencies. This, on the other hand, is going through a bit more processing. There's this weird EQ shape which I've done after I've distorted it. So there's again, an instance of trash without it sounds like this. So it gives it some balls. And then when you pre EQ it and I tell him which frequency I want to address. This is one when you distort pre EQ and post-acute fulfill completely different roles. So this pre-acute told the distortion, this is what I want you to saturate. And the post-acute kinda helped me shape it more. In other words, pre cue is behavior. And post-acute helps you shape the tone of that character that you created. I just wanted more weight in the low frequencies than this transects, which helps with accentuating transients. Let's go to the lead vocals and then we'll talk about processing of the backing vocals as well. As I mentioned in the arrangement stage, the lead vocal is spread out over four different channels, which are processed differently. Let's talk about the clean 1 first, the source isn't too good, to be honest, was poorly recorded due to various circumstances and alone. It sounds like this holy infinity all around us like heaven. 24-seven of questions. There's a lot of low mids. It was really hard to deal with these vocals. She has a very high voice, so I filtered a lot of the low frequencies and then started a dressing and pinpointing. Where are these frequencies that are cough, nasal, low nasal kind of frequencies. So these are the frequencies I've attenuated holding infinity all around us like heaven, 24-seven of questions, 0.05 per cent. And then a multiband compressor that is a bit more adaptive and is reacting to the Loman and bass frequencies. Let's listen to this holy infinity. You'll around us like heaven 21st, holding infinity all around us like heaven, 24-seven of question, really gentle, some more attenuation. And then I'm pushing the high frequencies but I haven't dynamic. So whenever she has an S or Sure sound, then it attenuates it wholly infinity all around us like heaven. 24-seven of questions, 0.05 per cent answers Wholly Moly holy molly and my soda. Then there's this altar boy which is also automated and comes in and out. And we'll listen to this when it arrives. More AQ filtering out these nasty low frequencies holding infinity all around us like Kevin, 24-seven of questions, 0.05 per cent answers. There's this dynamic EQ, which is again, a bit more alive and reacts to things only when they appear and not constantly. So let's AB no processing. And until this point, only infinity all around us like heaven, 247, No questions. 0.05 per cent answers, holy, holy infinity, all around us like heaven. 24-seven of questions, 0.05 per cent answer is holding, to be honest, if it was recorded better, I wouldn't have needed to do all this processing, but let's go on and see how I've further addressed this. Then there's this LA to LA, which again is a compressor, but it has a really cool tone. Holy infinity all around us like heaven. 24-seven of questions, 0.05 per cent answers, barely doing anything doesn't even reach one dB attenuation, then there's this distortion, which if you look at it, you see it's pretty wild, but actually the mix knob is down all the way and it's automated whenever I wanted to kind of push a sentence forward, I automated this and distorted the vocals and other EQ. This is automated as well. This goes up and down during the song and gives her a lot of character and the ester because her S's are hard, holy infinity all around us like Kevin, 24-seven of questions, 0.05% answer is holy moly in certain points on the verge of listening, but it's not. Then there's this whole bus in which all these vocals are routed to and just compressing it a bit within 1176, then there's some more overall EQ, mainly addressing the S's and these frequencies which I wonder how they sound. Yeah, unpleasant. And finally, sooth, which is a great EQ, which addresses resonant frequencies which come and go, holy infinity you will around us like Kevin, 24-seven of questions, 0.05 per cent answer is holding. It really helps keep her and check with the backing vocals. It's very, very simple as either telephone filters are just basic filtering. I just didn't want them to overpower the mix in any way. And different vocals received different kind of care. They each have a different kind of sonic texture and that helps the listener focus. The fact that they are different from the lead vocal just makes it easier to follow. So there's very, very distorted kind of ad libs. And then there are the more cleaner ad libs. And then there's the shout. This is just us shouting the same thing four times. I just doubled the pandemic left and right and bust it to one bus without processing, it sounds like this. Then I've put some EQ, some distortion, very, very mild molecule. And then it sounded like this. Just attitude and sent it to reverb, chamber reverb and delay. And that made it sound like this. Pretty simple. All of these elements in my template are going to stems. Let's say there are 1012 drum tracks. I follow them down slowly to a drum bus which feeds the stem. And the same goes to every instrument on. So there's the drum stem based on harmony stamp strings, backing vocals, vocals, and vocal effects. Not everyone is being processed. Let's just give a listen to the drums and see how the stem processing, which is a bit of saturation, just a bit of filtering of the sub frequencies. And the filtering of the high frequencies affects the drums stem feels weak and unmixed before. And you can see that saturation is a really, really strong tool. I use it more for what it's doing on the dynamics than to actually receive audible saturation. But more than anything, the cool thing I want to show here is what this gentle filtering does to the whole track. It doesn't go over 1.5 dB. It is gentle, but it actually is doing quite a lot. This attenuation really, really packs it all together. And something amazing I've learned from mixing this song is what impact the low mids have over the field of proximity of an instrument for the listener, this kinda makes the whole kit feel closer than the base stem has a bit of saturation. And I'm filtering all the sub sub frequencies and leaving them for the bass drum. This adds some character and attitude. Same goes to the guitar stems and on the vocals, there's a bit of EQ, a tiny bit at 18 k, a tiny bit at 2.5. And I'm using the pre to circuitry. This is introducing preamp saturation and then some more adjustments. Let's say be this finity all around us like Kevin 24-seven of questions. Finity all around us like Kevin 24-seven of questions. Just a bit brighter and more exciting. And then something pretty cool I like doing is parallel reverb to the whole mix and parallel compression usually, but in this case, I'm actually parallel distorting the whole mix. Let's listen to the reverb below around us like Kevin winning for seven of questions. Maybe you'll around us like Kevin winning for seven of questions. Holy moly. I like what this is doing. It's pretty subtle. Sometimes works, sometimes it doesn't, I don't always use this and then there's this nice saturated. And let's listen with and without the yellow around us like Kevin winning for 7.05. All around us like Kevin. Kevin, no questions. I really like what this is doing to the snare. It's kinda adding a lot of weight to the whole mix. This eventually sums up to this mixed bus which is actually already mastering. And this is pretty much it. 41. What you Give - (Get Familiar With The Song): Slow down. You will find yourself too often. They don't give it down. The left side because they don't care. What they know is pretty bad. As long as we're still, they'll never find writing on the walls. Just care what you give. Slow down your heart. You will find yourself too often. The fastest, don't bring it back. You will find there is nothing that's violence or this place. It's only doing something wrong, not just Christians. I'm just so confused. You just care what you hear. 42. What You Give - Mix Overview: Now that we are familiar with the arrangement, we can dive into the mix, I will mention again that my goal is to expose you to tools and techniques. So you might see plugins that you are not familiar with. The names of the plug-ins are mentioned in the segment and you can look them up if any of them catch your attention. As I mentioned, this is the kick of the song, but in the intro, I have a different sounding kick, which is sent to a pretty long reverb. And I filtered the low frequencies out of because I didn't want it overpowering the intro. And it sounds like this. The processing I've done here is slight saturation using the Omega transformer and I filtered the low frequencies. This channel does not go through any processing by itself, but it is routed to a bus. There is side-chain compression from the snare, and I'm slightly attenuating the sub frequencies. Let's listen how this sounds without this EQ. It just feels a bit more balanced. These sub frequencies will just overpower your master bus and eventually will require more limiting when you master. So I just decided to do this in the mixing stage. The reason there's side-chain compression coming from the snare is because the base term is playing four on the floor. When the snare hits, it is just not so necessary to have all this weight together. Let's listen to the snares with the kick. Now you might not notice a big difference when you look at this. But if we jump to our master bus and remove the limiter, notice where peak levels we reach when the such and compression is bypassed. So we have minus 16.6 and now that we have the side-chain compression, so we've gained a dB and a half or so of volume. And there's no sense of compression because the snare is the center of attention. This will result with less compression coming from the limiter in the mastering stage and just be more pleasant. I do think about mastering when I'm mixing, especially in electronic productions. Let's talk about the snares. We have two samples. The first sounds like this, and the second sounds like this. You can notice that the first one is our main snare and it's electronic. And the second one kinda brings in the rattle of the snares on an acoustic drum. And I'll show you how I got there. The original sample snare sounds like this. I've saturated it using the Omega transformer model n, which emulates a knave transformer. This gives it a lot of weight and unnecessary weight. So I've used this EQ, kind of balanced the fundamental frequency of the snare, and that apparently resulted with just too much high frequencies. So I've filtered the high frequencies a bit. Let's listen to what this is doing. Listening to this and solo, you might think to yourself, he has ruined the snare. Well, in a way you are right, but you are very, very wrong because no one listens to your mix and solo. I have the bass drum playing and the role that snare plays in the context of the whole song does not require all these low frequencies. After this EQ, I've inserted another model and saturate or which kind of focuses the energy from higher meds to slightly less higher mins and compresses it even more. With the second scenario, I'm doing a bit more violent processing, unprocessed, it sounds like this. And processed. It sounds like this. I'll explain why in a minute, but before, I'll explain how first thing I've done is to attenuate the high frequencies and this low mood. Then I've slightly distorted it again with the omega n. I'm assuming I bought the plugin around the time I was mixing the song. So I've just experimented with it as much as I could. As I mentioned in the last instance of the firstName, it kind of focuses the energy a bit lower in the frequency spectrum. Next up, we have the amazing 1176. I'm using all buttons, N mode, which is extreme compression, and pushing the input extremely. So it does give it a lot of snap, but more than anything, it just keeps the snare rattle longer and creates this long the k that i then balanced out using a transient designer. So what I've done here is to lower the sustained because it was just too long and did not serve me rhythmically. Let's listen to the two channels together and, or bypass and on bypass the two dynamic processes. You can just hear these somewhat natural high frequencies coming. And the last thing to note is that the first snare is being sent to Plate Reverb, which is not too short and then filtered or the NICU, I'm cutting off 130 downwards, the marching snare. And the C part is something I programmed with Midea and unprocessed. It sounds like this. With processing. You may recall that the beginning of the Seaport is very open and there are not many elements. So this long reverb is kind of okay, but if the power was more dense, I wouldn't have sent it to such a long reverb in this amount. The first thing I'm doing is to cut off all the frequencies and attenuate some low mids. Then I'm sorting it using trash. Trash is a pretty cool distortion unit by isotope. There's the trash module which is saturation. Then I'm filtering it. I'm adding a bit of high frequencies and I'm using a convolution reverb. It's as if the sound was recorded in a helmet. And then I'm adding some delay. Let's listen to what trash is doing. So it's adding a lot of excitement and the distortion kinda compresses it and the ghost notes or just more apparent in the mix. Then I'm starting it again with the commutator. I'm using the pentose style and I'm distorting it extremely, but dialing down the mix knob. This brings up the room sound from the recording that I had to go through a few stages saturation. And generally I'm just degrading their audio quality and I'll explain why in a sec. Originally they sound like this. The first plugin I'm adding is saturated or by an brainy, ever so slightly saturating it. It just makes it a bit darker. Next up, I'm using soft tubes, tape and I find myself going to be quite a lot. It also chops off the trends in a bit more and brings the decay slowly closer to the pigs. Finally, I'm filtering off all the high frequencies. And we'll understand why I did this only in contexts. So let's listen to this. So these piercing high frequencies don't really serve the whole vintage feel I was going for. They just pick too much. And with this processing, they can sit in the mix better. I try not soloing stuff because it can really draw you away from what it is that you need to do. And sometimes in mixing, you just degrade something to a certain extent for it to leave room for other things. As for the crash, I'm just filtering few resonating frequencies and then the reverse snare and pitch shifting up a fifth and using this plugin to create something like a reverb. Notice how the sound without this login and width. I don't really know how and why this is doing this, but it is. Let's now talk about the bass guitar, which is going through some very interesting processing. When we were working on this track, we rented a studio that had a horrible base in it. And I've actually needed to auto tune it in the mixing stage because it would just not stay in tune for long. And after the auto tune, I've inserted two base and emulations. Let's see what each does. This is only the first. And here's with the second. So it gave it more attitude and kind of focus, the mid-range attack of the pick where I wanted it. And then what I've done is to route this base channel into these auxiliary sends. The first one, you might have noticed, I'm calling it base low and filtering all the high frequencies up until 150 hertz. The reason I'm doing this is because I want my bass frequencies centered and in control and clean and focused. And I knew I wanted to distort the high frequencies and have them on the sides. After this filter, I've put multiband compressor, which reacts a certain notes more than others and keeps my base and check, let's listen to this. You might have noticed that without it, the base frequencies were just too dynamic. And as I mentioned, I wanted them very focused. And lastly, I've just attenuated this frequency which currently bugged me. Then I have to high channels. The first is for the courses and the other four versus the reason is that I've created a bass sound in the beginning of the song that worked beautifully. And then in the verse is it just overpowered and grab too much attention? And it made it hard to listen to the singer. So let's start off with the one in the courses. The first thing I've done is to filter the low frequencies and to filter this range, which is the pick. Apparently I did not like it too much, but you can see it's a dynamic EQ and it reacts to the source whenever something just goes overboard. So the reason this filter is not at 150 hertz is apparently next plugin, which is an emulation of the rat distortion. I'm assuming that what happened is that I played around with the filter to shape what the distortion is actually reacting to that resulted with this sound. Then I've inserted sews, which is a great plug-in that helps manage resonances that are unpleasant but not static. Without it's just out of control. It's a bit too wild. And after I've inserted another EQ, this time the filter is placed around the same frequency in which I filter the high frequencies and the low channel and continued shaping the bass sound. So again, you might be thinking to yourself, why is he ruining the source? The reason is that I have the low channel, which contains all the low frequencies that I want. I don't need more low frequencies in the channel. I wanted only high frequencies. And the last thing I'm doing is to make this mono source stereo using Microsoft. And with the best channel. This division of the bass guitar is a really cool trick. And let's go over what I'm doing at the Vs. Just a bit more mild in this channel, the only variation is that there is slightly more distortion than the first one, but there's a lot more filtering going on. So let's high frequencies that are harsh and are taking the space from the lead singer, then the BQ, another EQ, which is filtering more of the low frequencies. And Microsoft, the last thing to note about the bass guitars that it's sent to the bx 28th spring reverb, which is automated, and it only appears in the end of the song. So when the song ends, it ends with the bass guitar decaying out. Notice what happens in this fader. Spring reverb immediately creates this vintage feel and it's sorted this production beautifully. The bass synth has nothing going on except for this filter and slight attenuation in 187 hertz. There is no need to do anything extreme. Don't bless you, do the better. That being said, do whatever you need in order to get where you want to go. Now, the guitars, which are going through very, very extreme processing, these guitars were recorded directly into the audio interface and without any processing sound like this. I know. First thing I've added here is guitar rig, which has first module, the orange AMP and Marshall cabinet, which created this sound. Then a preset of trash, which is very, very violent, going through another convolution reverb and some compression and delay. Then why not another distortion. Side note, this is clipping, which is another form of distortion. There's practically for degrees of distortion in this processing chain. And lastly, spring reverb, because again, we're going for a vintage feel and it suits this beautifully. Now we're talking next up. I've routed these two guitars to a bus, and I've EQ did. I've added some high mid and filtered all the sub frequencies because when you extremely saturated Something, sometimes it results with this. These are frequencies are just taking space from my overall levels and are not necessary or audible. So I've kicked them out after this EQ that I've used for tonal shaping, I'm using this EQ, which I've automated to create this effect. And the last thing that I'm doing is to automate the mute button in order to not have the decay of the reverb and have this vacuum effect. Automation is a very important part of a good mix. It gives life to mixes, it creates more dynamic mixes and it's something you should invest some time and moving on to the guitar, which actually sounds like a guitar. It's nice but somewhat two-dimensional. So I've inserted this instance of a 610 emulation. 610 is a legendary console built by Bill Putnam, the founder of Universal Audio. This console has tube circuitry and although I did not work with the analog console, I love the emulation. They did no matter how close or far it is from the original thing. What I'm doing here is a pretty cool trick in which I push a lot of level into the console and then I lower the output so I'm driving or overdrive the whole circuitry and creating some distortion. I really like, Let's listen to this. I think they've done a really good job in emulating tube distortion. It kinda feels a bit more three-dimensional and brings the pitch sound a bit further. And then I'm compressing the guitar mildly with this 1176 emulation, two to one ratio. So this enhances the pitch sound even more. But something interesting I want to point out is that I've dialed the release to a setting that stays just enough for the sound not to feel choked, but it does attenuate this annoying low, mid-frequency that just sustains too long. Try noticing this. It's very subtle. You might have noticed that, you might have not noticed it, but I really like what this is doing. Finally, I'm sending it to a plate reverb which pushes it backwards in the mix. If this was the only truck in the mix, I would prefer dry, but it doesn't have a very important role. So I pushed it to the back in order to leave space for whatever I wanted in the front of the mics. Moving on to the, since the jumbo is going through this EQ, which is essentially a part of the whole agenda you've already met in the drums, which is making things a bit low fi. What I'm assuming that happened here is that I started off rolling the high frequencies with the high shelf, and that resulted with a few frequencies picking out. So I needed to attenuate these a bit more surgically. So as I assumed, it just degrades it and makes it a bit more low-fi, but in a controlled way that does not pierce our ears. Then I'm sending it to a digital reverb. They're all digital, but this is an emulation of the first digital reverb, actually the EMT to 50. I've set it to decay over a pretty long time. Then I'm sending it to course and another course, essentially, which is macro shifts, let's listen to it dry and then see what the effects at. Grand juror, bigger, wider, and more vintage you the B3, I'm using the soft tube tape again, a mount for on the base setting, but this time I've set it to 15 IPS, then I'm enqueuing it. Nothing much. Now, very dramatic, just mild shaping then that ambient lead synth, I'm using Sketch case2, which is an incredible plug-in, except for just adding this cassette kind of tone, I'm adding some Wow, Which is, which is relatively slow fluctuations and pitch. And then I'm filtering all the high frequencies of course. So when it was unprocessed with all these high frequencies is just felt thin. And four, and having all the high-frequency is filtered, the sound a bit more focused in the mid-range, made it feel a bit closer. This sound is sent to a ping pong delay set to eighth notes. I've chosen the memory men style Santos a cobol has a lot of styles to choose from. And I do search. I tried being very accurate with my folks Joyce, the line and the last chorus is also going through tape and is just being extremely filtered both in the highs and lows. Let's listen to this unprocessed and processed. And so again, you can see that the mid-range has a lot to do with how close or far things feel before the filtering it kind of feels for and when it's filtered just feels closer and more powerful. It's not necessarily the case always, but in this case, it definitely is the line in the first verse. I also have extremely filter. I also filter these frequencies which were apparently resonating and annoying. And then I've inserted trash, which I use to gently distort the sound with tape saturation, then filter the low frequencies and slightly compressed. Let's hear this source. Why cooler? Yeah, high frequencies are overrated. Then we have one channel unprocessed. I'm just sending it to the two courses. There's no need to do anything. Don't. I mean, I am doing some extreme processing here, but it's not because I want to process this because I felt the source needed it when I don't feel that I don't do anything. Moving on to the next arpeggiator, I'm using Sketch cassette again for its well and just the sound of the cassette filtering the low frequencies, slightly attenuating the high frequencies, or not slightly at all actually widening it using the micro shift. So again, I've just ruined the source in order to leave room for other things that happen. Jumping to the backing vocals, Let's talk about the formant altered pair. The first thing I've done is to filter the low frequencies and bursts some high mids into the unit than just dial down the formants to almost minus five. That resulted with this. Then pushed some more high mids and inserted spring reverb. This put the source in space and it really flat, or is that done some compression using the 1176? Still be dynamic but controlled than I've inserted and finished off with rolling all the low frequencies off. As for the distorted vocals, I'm using sand XAMPP, which is a beautiful stock plugin that comes with Pro Tools. It has the premium saturation going on. And then these three are low frequency distortion, mid frequency distortion, and high-frequency distortion, then an overall drive and then low and high EQ. This has a very specific tone and it's beautiful and it's followed with a DSLR because the inch and S sounds just became too harsh. And let's see what this is doing. These were sent to a plate reverb. Which is pretty long, and that is it, just tone shaping and control. Then we have the vocals shouting, I just care what you give, which are processed with very light EQ, they more than 2.4 dB attenuation. Then we have trash distorting the bejesus out of them, slowly filtering the low frequencies and pushing the high frequencies and adding some delay. And finally some the S in to make sure that the distortion does not pull our ears off. Lastly, it's sent to a ping-pong delay with quarter note triplets and a lot of feedback. So without any processing, it sounds like this gesture, what you gave with the tonal shaping. It sounds like this. There is some delay in the trash module to remind you. And with a ping pong delay, let's listen to how this sounds and contexts. It's not horribly obvious and frontal. It's put right in place. Let's talk about the lead vocal. There's a big secret in the approach I came with to this vocal. We had a good performance with horrible quality recording and I needed to choose if I'm coming with the approach of making it as hi-fi and big as I can. Or on the contrary, goal for a low fi kind of feel. And instead of compensating, just flatter what it is that I do have and that is the approach I was going for. So the first thing that I've done was to take care of the proximity effect that was in the recording. There were just low mids everywhere. It was really hard to deal with, to be honest, after I balanced this, I've inserted another instance of rats or raw. Let's listen to what this is doing. Slow down your heart beat is the city of the dead. You will find yourself alone too often. This makes it obvious that the low finance is intentional and it's more of a stylistic choice than just allows the recording, which is not something that you want people to feel when they're listening to your songs after the distortion, I'm just slightly attenuating these frequencies, which are bugging me, and these high frequencies which are just harsh. And also the low frequencies which are just not necessarily slow down your heart. Bead is the city of the dead. You will find yourself alone too often. This keeps the distortion from piercing our ears. And additionally, I'm using sooth to even control this. Further, I've dialed down the mix to 81%. Slow down your heart beat is the city of the dead. You will find yourself well-known too often. Controlled balance. It is distorted, but it's not horrible. And let's listen to this in context and see how it interacts with the other sound. Slow down your heart. You will find yourself too often. They don't give a ****. Because it fits the scenery, it fits the text, it fits the melody, and it works great. Now, I want to talk about automation. Here you can see that the harmonic instruments go through quite a lot of automation. This is volume automation. Specifically, after I'm done shaping the tone of the instruments in the mix, I play the song from the beginning, balances. And then as the song plays, I try noticing what pokes out and what just feel static and need some more movement. This volume automation are things that I invest quite a lot of time. And as you can see, I'm going to very fine detail here to make sure that the vocal is in place throughout the whole song. And there are not essays or peas that are just popping out and just annoying my listeners. I also automate effect sense. You can see that here I have this reverb throw. Let's listen to this writing on the walls. I'm just so confused. I try noticing where either the texts or the melody requires more tension and then I can send it either to a river, but delay and give them mix more life. My whole mix is summed down to these stems. There's the jumps down based on guitar stem, etc. Here, if I feel there's a need, I do mastering. And in this mix, I did not do much. There's just minor E Q on the base and some EQ on the vocals, slightly shaping it. And actually, the last thing that is happening before I'm x goes through my mastering chain is these two parallel channels that are doing parallel processing for the whole mix. So I have a river parallel, which I'm not using in this mix. I sometimes use, I sometimes don't. There's this parallel channel, which are usually used for parallel compression, but in this case, I used for parallel distortion. It's very, very subtle. I feel that it kinda keeps things a bit more in place. It adds a bit of energy. And this is how I've mixed what you give. I hope this wasn't so forth for you. Can actually download this song with all its files and mix it yourself along with songs from other genres you can practice on. Let me know if you have any questions and I'll see you in the mastering sediment. 43. SkillShaer Outro: Hey guys, I hope this class was insightful for you. It's really important for me to note that knowledge can only bring you so far. And in the end of the day, practice is what will take you to the next level and make you a professional engineer. In my full course, start producing music.com. Besides the exercise files, you will receive multi-track folders of songs from different genres, along with access to a private community in which you can share your progress and receive feedback for your work. There are also free sample packs and I constantly released new content to keep the learning relevant and up-to-date. And so when you sign up, beyond all the current content, which is over 115 lessons, you'll receive all future content with no additional cost in the business chapter, I share insights as a practicing professional with the intention of saving your mistakes that I've done in order to learn these lessons myself. This course is a well-organized, condensed and super practical training that will help you develop your music production skills and shorten your learning curve on your lifelong journey as a musician and music producer, makes sure you go over to start producing music.com to learn more about the ultimate online training. And I'll see you.