My approach to tackling projects, whether for independent artists or from record labels seeking music tracks for commercial release, involves several stages prior to the music production process.
The initial step typically involves engaging in thorough discussions with the client to understand their vision, expectations, and any specific project guidelines. This step is crucial for providing direction and ensuring a clear understanding of the project requirements.
With all necessary information gathered from the client, I proceed to conceptualize the track for production. If I am familiar with the chosen genre, I dive straight into production. However, if the genre or project is new to me, I dedicate time to studying the genre to ensure my work aligns with the client's needs.
For my scenario of being approached by a record label to work on a music track for commercial release, I have chosen the genre Retrowave/Synthwave. This genre enables me to demonstrate my in-depth skill and understanding of advanced music production techniques, including working with audio, synthesis, sampling, mixing, and mastering.
Melody - Subtractive Synthesis
I began the production process by crafting an improvised melodic structure which would later be supported by chords and a counter melody. I would experiment with various combination of note structures on the fly allowing for the organic emergence of melodic patterns. Once I have found a melodic pattern pleasing to my ears I would follow the associated notes within the melody to enable me to identify the key and also to discern which notes complement each other effectively. This is one method of approach I use to provide a “body” or solid framework for building the song and also for inspiration on making my music unique. Whilst figuring out the melody I would begin designing the sound for the melody.
The sound I would design for the melody would need to fit the genre as sound selection is crucial in production and usually identifies a genres sonic character. I chose the jun-6 for this task to demonstrate my use of subtractive synthesis and to form my melody.
I began forming my sounds shape with the envelope settings. I wanted a sound which when pressed it wouldn't be sharp initially and would duck at the beginning and when it would peak it would drop in loudness but on release it would still go on. this lead me to set the envelope with a slightly slow attack and slight decay with moderate sustain and a high release. I then would begin to filter out some of the high frequencies around 450hz so the sound is more darker sounding than bright. I would compensate this by adding some resonance in my sound. I then would set how much my envelope settings would affect my filter with modulation and would reverse its effect so it makes a swelling effect when a note is played. (shown in video demonstration).
Following on from this I would switch to a square wave as the saw was too sharp sounding for my taste and the square gave a more soothing bell sound. I would have to revisit my settings as the sounds volume would be lower and this had lead me to the idea of creating space in my sound by adding a slight amount of delay effect on my sound and spreading the delay to the sides more. I would also set the delay so the timing would be offset to my original sound.
i then chose to modulate my sound with the 1st LFO and would decrease the rate after a short listen to a slower rate which I then got to the target sound pleasing to my ears. it gave me that early 90's SEGA genesis game sound commonly heard in the early arcade games full of subtractive and FM synthesis sounds as many famous producers in the gaming industry such as Yuzo Koshiro would use.
I then would work on a second sound source to add to my original and I initially wanted to add a sub as my second sound source but after some playing around I thought to use noise as my second source and would go about in shaping the sound with an envelope so that the volume starts low and swells louder as it progresses. the setting would be a very slow attack with slight decay and sustain ending with a lot of release. This had formed into an ocean kind of sound or the sea. it would accompany my 1st sound and would be like a tail to the overall sound meaning when a note is pressed the initial sound would be heard with the delay effect and the 2nd sound (ocean would follow after by swelling in loudness).
I would add some sub and lower the volume so it doesn't clash with the 1st sound created.
This would be the finishing stage of my sound design as I have reached a target point and would spend the time refining my sound created.
I would link my 2nd LFO to my delay effects feedback and also to my filters frequency setting for some movement so everything gels together and sounds more unified rather with sections sounding still.
I would also set how the sound is played when I press my MIDI keyboard so that I get a more better sound rather than relying on a random velocity which in turn would affect my sound as I needed a consistent sound which in turn makes it easy to play and perform with. I would also link the velocity of the keyboard to my filters cut-off so that when I press harder on the keyboard the filter is affected. I also would adjust the keyboards portamento setting to add a glide to the sound when played.
once finished i would listen to both sounds indiviually and combined to see if i am overall happy with it..
I would then begin to compose my melody by filling it with associated notes to make a set of chord progressions for my starting of the track. I would also improvise a counter melody to go with my main melody/chord. Once I am happy with the improvised counter melody I would also record this and begin editing it by quantizing my played notes to be on time and adjusting wrongly played notes.
Following on from my melody I would create another sound which initially started off as another melody I was thinking to layer with my starting melody but I had an idea for movement with this sound and applied a software plugin (Filtershaper) which has 2 filter engines with time based adjustments, various filter shapes and the flexibility to create amazing sounding patterns and movement in a sound. my 1st filter would be a filter cut-off and shaped so that the amplitude gets louder and then quieter in time. I set the timing interval of the 1st filter in sync with my tempo and to 1/8 so the filter effect happens at a fast rate causing a ducking effect. For my 2nd filter I applied again the time interval of 1/8 and in sync with my tempo. The filters shape was a sharp cut-off so that when its working alongside the 1st filter my ducking effect would have a more sudden ending tail whilst the start would be smooth.
Bass - Sampling
My next idea was to add a bassline to my melody and I had chosen to demonstrate multi sampling in its early forms and form a bassline by layering samples. I chose to use one of the early pioneers in sampling which was the E-Mu systems Emulator 2 released in 1984. Used in countless hits (as seen in the video) ranging from Phil Collins, simple minds and enigma to name a few.
To briefly explain its use, on the left side is a column of boxes in which audio samples can be loaded into. Once a sample is loaded in (in my case I chose 2 samples CS30 Bass) I am able to manipulate the samples behaviour and characteristics with the shown filters, LFO's, Velocity settings, Looping functions and effects to name a few. I wanted my bass thick so I added 2 of the samples and transposed one of the samples to a lower octave giving a fuller bass sound. I would set my 1st sample to have a loop at the end so if I held a note on my midi keyboard the sample will loop according to my chosen position. I am also able to setup the loop function so that it loops either from the beginning, In reverse, or it can bounce between the two loops. I would set the velocity of both samples to its highest settings so that the audio would not change on how hard I press a note on my MIDI keyboard.
Drums - Audio
My next task was to begin adding drums for a groove and I chose to use my hardware MPC Native
Instruments Maschine which is similar to the Akai but the feel of the pads are different and the velocity sensitivity is different. I had samples of drums I have recorded and layered onto the pads of the MPC and began improvising a groove with a Closed Hi-hat and a clap which had a Retwowave/Synthwave feel/sound. I then would use a Function on the MPC which makes notes I press play in time by syncing to my projects tempo (quantised) and i would perform and record a groove. I would create a new group on my MPC to layer more drum sounds and I would use "retro pop" a drums Kontakt library of drum sounds. I would just layer another hi-hat and a kick on top of my previous HI hat and clap. Following from this I would layer a crunchy sounding kick and another lower sounding Hi-hat with a long tail. Lastly clashing cymbals to give more impact to the layered drums at a certain point in time. My aim later in the mix stage would be to tame all the transients from the hi-hats and kick with compression and using gates and grouping.
Arrangement +FX - Audio
I now had substance to my composition and was able to begin arranging a structure to my composition. I usually get ideas during this stage such as effects for the intro or elements to add or remove. I had the idea of a tape stop effect prior to the start of the first verse section of my arrangement. I achieved this by slicing the audio prior to the start of the verse section and removing the audio. I then would place an effect plugin on my master bus and design my tape stop effect. This technique allows all the audio to be affected by tape stop effect and I am able to control when the effect starts on my audio by adding automation to the tape effect.
With my 1st half of the song structured I began to think on the idea of having a break in the song and bringing in a different arrangement of my composition with new elements added (similar to how Retrowave/Synthwave music has switch ups).
I would slice all audio and place a slice of the clash I created in my drums section to be as an audible signal of the transition I want for my next arrangement. I then would create a different melody with my subtractive synth patch I created at the beginning and arrange and layer the drums in a different pattern.
Melody 2 - Additive Synth
My next task was to add something to my second arrangement and I also wanted to demonstrate additive synthesis for my assignment task. I chose to go back in time and use one of the most famous early digital synthesiser/sampling systems the synclavier ii. Again used in countless hits and full of powerful capabilities ahead of its time. Above is a short video clip salvaged from my fathers old VHS tape of when it was publicised and introduced to the media, schools demonstrating its possibilities at that time.
(Full Documentary added into to files for viewing)
To demonstrate my knowledge and understanding on additive synthesis and FM synthesis I will firstly explain the functions on the synclavier and a visual demonstration in the form of an EQ I placed to visualise the additive content/harmonics introduced.
(whilst this is a software emulation of the synclavier the use has been designed similar to the hardware and what would be seen on the computer side of the original system.
The way the synclavier worked with audio was in the form of several sound engines known as "Partials" which when combined would create a single sound known as the "timbre". The partials themselves are created by stacking modules called "frames". An Overview of the building block would be the frames which when combined will make partials and when the partials are combined we make the timbre. (a three level hierarchy in sound design).
if we view the monitor to see the synclavier ii's engines we can see
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12 partials listed on the left side in a column.
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2 blue boxes below (one named Carrier and the other Modulator) which both have bar graphs and a sine wave graph (Harmonic editor)
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A timeline at the top which is the frame and above the timeline we have a menu bar for different settings
With a visual EQ I am able to see that when I play a note we can see the frequency information on the visual EQ.
As I start to introduce more bars in the carrier box we can see harmonic content being added into the sound of partial 1. This is because the bar graphs in the Carrier and modulation box represents the harmonic content of the sound we are playing. the numbers 1 to 24 is the number of the harmonic of the sound meaning 1 is the fundamental and 24 is the 24th harmonic.
With this knowledge if I increase the bar in the graph the I will be introducing or adding harmonic content to my sound as seen in the EQ. The sine waveform display below also changes as i am adding harmonic content and this is because a sine wave itself is a single frequency meaning it has no harmonics just a fundamental frequency hence why there is only 1 bar show with the sine wave but the waveform changes as I introduce harmonics in the graph editor.
we can also see that when I switch the harmonic graph to a sawtooth; the harmonic content in the graph contains all integer harmonics showing each harmonics amplitude decreasing proportionally to its fundamental frequency i.e. the 2nd harmonic is half the amplitude of the fundamental, the 3rd is one-third and the 4th one-fourth. Overall we are able to do our additive synthesis here and also shape our sound
The right side of the graph (Modulator) is where we can do our FM synthesis by using this waveform to affect/modulate the carrier waveform. The modulator section works similar to the Carrier side in where we would form our waveform by adding harmonics but then we would introduce the amount we want to modulate our carrier in the mixer section. As demonstrated in the video this introduces the typical metallic FM sound with FM synthesis.
I began sound design by firstly creating 2 frames for my first sound (partial 1). I would set it that after pressing a note the 2nd frame will take effect roughly 2 seconds after playing a note.
I then would draw a random pattern on the phase setting of the harmonic content on my 2nd frame so that when I played a note the sound changes in a warped fashion. (you can also see in the visual EQ how the audio is behaving by its movement in amplitude. you can also see purple visualizers on the synclavier interface showing the behaviour over time. I then would begin working on my 2nd partial and would make it a sawtooth wave and would transpose it higher. I would then create a frame on the timeline for partial 2 similar to partial 1 BUT I would remove the 1st frame so that we only hear the tail of partial 2.
This created a swelling sound which I liked and after some improvisation on my arrangement I loved how it sounded as a sub!! this was due to how it behaved as a sub having a lot of low end but also introduces audio in the mid high frequency range which is great to help when mixing so that its audible on smaller speakers. I would arrange the pattern and review for final touches.
Melody 3 - FM Synthesis + Matrix Routing
Operator 1
My next task was to add a counter melody to my second arrangement and I would use this task to demonstrate my knowledge and use of FM synthesis further and matrix routing capabilities. For this task I decided to use FL studio's Powerful Factory plugin Sytrus which has 6 operators, 3 filters, FX and routing capabilities. Each operator can be seen as a subtractive and additive synthesiser or even as a module alone but the overall function of Sytrus is as an FM synthesizer with routing which allows a user to modulate operators with each other to create complex sounds.
I began by shaping operator 1 to a starting point I could work with experimenting with various waveforms. I then would remove clicks and pops in the audio and begin forming the shape of the sound with an envelope. Sytrus has routing capabilities in which I can route the envelope to either the pan, volume, modulation, pitch, phase, dampening or oscillation functions. I routed the envelope to the volume and would transpose the sound to a lower octave and shape the envelope to have a long release. To counter the overlapping sounds I would set my polyphonic setting of my keyboard to 1 so only 1 note can be played at a time. Following from this I would reshape my waveform to have resonance so it sounded metallic. I would also start introducing operator 2 just to see how it would sound and I was AMAZED by the sound formed which was similar to the early 90's games produced with synthesis. This signature sound is familiar in many early MS-DOS PC video games ranging from Simon the sorcerer, Monkey island up to big hits on early consoles such as Final Fantasy7.
Operator 2
With operator 1 out the way I began to work on operator 2 by firstly selecting the waveform and shaping my sound with the envelope. I would also introduce operator 1 by turning the yellow knob on the out setting on the far top right of Sytrus. I would do this to ensure whatever I create aligns with my 1st operator. I would then go into the filter section of Sytrus (GLOBAL FILTER NOT THE OPERATORS FILTERS) and begin adjusting the filter to shape all sounds. For me to hear the filters effect on my operators I need to route this in such a way that the audio of the operator passes into the filter first before the out stage. This can be achieved by using the routing matrix by turning on the filters knob on the column of the operator and then to the out stage as shown below for operator 2.
I would then begin shaping the envelope of the global filter routed into operator 2. I would also route the filters pan to an LFO to create movement from side to side when a note is held. the aim was to have slow movement in my sound so that when it comes to modulating the effect wont be drastic as I have learn to use FM modulation in subtle amounts if the maths isn't ones strength.
Routing Matrix
With both operators at a point I was happy with I began with FM synthesis by routing the operators to see the effect gained. I would begin routing operator 1 into operator 2 and turn the yellow knob (phase knob) to the right which in turn introduces a certain amount of operator 1 onto operator 2 which then goes out to audio enabling me to hear the frequency modulation caused by operator 1 as the modulator and operator 2 as the carrier. I would then route this also to the global FX column next to the out which I would set to a large reverb. I would also remove the dry audio of I am working with from the out and just hear the reverb or readjust my waveform to see what sound I would get.
After some testing I would keep only the reverb as the output for my sound rather then the dry sound as it sounded more smoother rather than sharp. I would also transpose operator 2 so that both operators have different tones
Overall I had operator 1 as the modulator and operator 2 as the carrier and this was going into the reverb in my FX panel. I would then begin to route operator 1 into my Global filter number 2 which was shaped with a slightly slow attack and short release. I would have this routed so that it runs parallel to my other sound by setting its routing separately and going out. (in a sense it would like a side chained effect).
Final sound
With my sound fully designed I went about layering the pattern I had in mind and how I would use the sound I designed. I would also arrange the 2nd arrangement of my song to a final stage.
Ambience - Wavetable synthesis
For a final touch I needed more sound information in my switch to the 2nd arrangement as it goes very quiet after the first arrangement having a lot of energy and sonic information. I would also use this task to demonstrate my use of wavetable synthesis using Arturia's Pigments which is a powerful synthesis capable of all forms of synthesis although strong on wavetable synthesis.
Pigments allows me to upload any sound to use as my waveform to work with although I am not that advanced working with complex waveforms so I stuck to the basic waveforms which can be selected as shown in the video. I am also able to "morph" the waveforms which blends the waveforms together giving me a sound in between i.e. a sound of a sine wave mixed with a triangle (I am able to blend this smoothly or just stick to a set waveform) I chose in between a sine and a triangle.
I would then add some frequency modulation into my sound and I would use Pink noise or Rumble to modulate my sound. I would use the ever so subtly as I wanted my sound to have a lo-fi vinyl wobble effect as it would sound very interesting to the ear. I would adjust it to a point I was happy with. I would also adjust my envelope so that my sound would have a longer release and slow attack which in turn created a soothing bell sound. I would then begin layering my sound to create a melody for my 2nd arrangement.
I would also create movement within my sound by routing the pan of 2 global filters on my sound to the envelope I made i.e. each to filter is on opposite side of each other and i.e. one filter left and the other right. I would automate this so that when the envelope comes to an end both the pans are returning to the centre. I wanted a delay affect so I would load up eventides H3000 plugin which is an emulation of the famous digital effects hardware they created. I would route the left and route audios to the delay patch and then to the output and i would then bring the mix halfway for a space sounding ping-pong delay. I would also chop and slice the audio to create an echoing effect at certain points of my melody as shown.
Lo-Fi Ending - Advanced Audio editing
To finish off my track I decided to create a Lo-fi style ending in which the track switches to a slowed down version sometimes its a reversed version.
To start this I would consolidate/bounce a section of my track and place the new section at the end of my track. I then would set this consolidated sample to be in sync with my projects tempo. Following from this I would create an automation on my project tempo and have it begin at the start of my consolidated/bounced audio. After setting a tempo I was happy with and drawing the automation I would begin working on my lead up to this Lo-Fi change of beat but adding a filter effect and a stutter effect prior to the change. I would place Akai software's Flex beat on my master channel and create an automation clip and draw an automation so that it only kicks in prior to the Lo-fi effect.
After a few listen throughs i was happy to say i had finished my task. Looking back and evaluating my practices and approach to this task I believe there was room for a lot of improvement on the instrumental side of my track as I had noticed I neglected some of the melodies in the 2nd half of my song and also I focused more on the drums due to my time management around this project. For future improvements I would ensure that I approach a task from a client with a more organised approach and to better arrange my timekeeping on a project.
Mixing
Before I would begin with a mix I would first ensure that I am organised as in the mixing process I believe organisation helps a lot and keeping things tidy whilst I feel in the the production process I like to be less critical. I would begin by naming, colour coding and arranging my session to suit me and the clients needs if there was a request. I would also have a Loudness analyser on my master chain at the end giving me information regarding the loudness and dynamics of my song and mix and a tonal balance indicator which shows my songs tonal information in reference to many other genres tonal information for reference. (my reference I set to pop being that my genre is close to pop)
To begin my mix I would start with my drums by grouping them and firstly gain staging the volume to a balanced level. I would also route them into a bus group naming it (Drum Bus). I would also pan some instruments to give each sound its own space and to help in stopping sounds clashing with each other. I also would spread some sounds in the stereo field more to give more spatial feel to some sounds. (Bass sounds I usually place as full mono to avoid problems in mixing the low end). My next step was to begin working on the drum bus by adding a Plugin which emulates the SSL 4000k Dynamic strip on the console (Antelope Audio's plugin). I would firstly begin compressing the bus track to control the overall dynamics of the bus/aux track as I had my heavy kick loud and some of the hi-hats were lower volume. I would set my threshold to a point in where the compressor would catch the peaks and I would see around 3 dB of gain reduction. I then would set my ratio usually starting at 4:1 and see if I'm not compressing the audio too much. I then would shape my overall sound with the attack and release. A slow attack and fast release would gives me a punchy and dynamic sound whilst a fast attack and slow release would give me an aggressive sustained sound which is perfect for my drums. I would then add and adjust the gate to tidy up the hihats as i had layered several hi-hats in my production with different tails. this helps keep my groove tidy and tight whilst my drums would be hard hitting and aggressive. I would later change my plugin emulation of the consoles dynamic strip to a different plugin (Plugin alliance Brainworx consoles) as this has several versions of SSL's console emulation and they are very good that SSL are proud to have their name endorsed with the companies emulation of their consoles. (all thanks to the TMT technology by Brainworx). i would first try the SSL 4000G consoles sound on my Drum bus and would later settle with the SSL 9000J console emulation as the characteristics of the dynamic section was amazing to my ears specifically the way the gate would tidy up my Hi-hats perfectly. I would later on critically listen and isolate any sounds in my drums that needed subtractive EQ work such as at one point my clap would have a clicking noise around 1k so I would use an EQ to isolate and remove the sound by raising the frequency band to pinpoint the frequency the noise rests at and then narrowing my EQs Q factor just to where the click noise is and lowering that section so that its not audible. My next task was to work on my snare and I wanted a Hard hitting snare like Michael Jacksons Hit song "Leave Me Alone". I would place a plugin called Drum shaper by XLN Audio which is a very great plugin for shaping or taming transients and was perfect for my needs. I would add some sustain to my kick and a medium attack. I would add a lot of "mojo" which is basically behind the scenes EQ which affects certain frequency bands at the twist of a knob. Every time I would make an adjustment or a change in the audio I would usually bypass the plugin or effect to see my results compared to before as it allows me to know if I must compensate in volume or other areas after my changes. Most plugins have things in place to help compensate the changes made such as "auto-gain" although it is good to hear and train yourself to know the differences and necessary decisions to make for compensation. I would place an optical compressor on my kick just to tame the peaks as it was too hard hitting.
My next idea was to begin setting up parallel compression for my drums as I wanted more thickness and body to the full drum sound for later when I begin adjusting the other instruments. My usual parallel compression technique for drums is based on the New York parallel compression technique which involves blending a heavily compressed audio signal with the original dry signal. I would firstly sidechain my drum bus to an empty track and then on the empty track I would firstly place a Fet compressor (1176 style) with a high ratio and a lot of compression so that the audio is getting crushed. I would use a fast attack and fast release to catch the transients and maintain punch and then following after this in the chain I would place an optical compressor (La-2a) and set it also high but just enough to let the transients through. I would then lower the volume down on the fader of the new York style compressed track and slowly raise it up to see how it sits with the dry track (My aim is to have the heavily compressed track to sit just under the dry track so that its like a support for the transients of the dry sound but not overtaking the dry sound. I would later add Baby audios New York compressor plugin in my new York style compression chain as it is great for a punchy characteristics on a drum bus.
Once this was done i would adjust the other sounds accordingly and adjust volumes where necessary. I would also take a break and come back and listen to my mix in reference to many environments using my Steven slate VSX Headphone and software which is a great "eco system" for mixing and referencing.
I would do a car test to check the behaviour of my low end and also a club test to see how the bass sounded in an open environment. I could also check on smaller devices or mono devices to see how my low end translates or if the high end is too excessive. I would save my 1st mix revision and would be ready to master.
Mastering
Prior to my mastering process begins I like to have a few things ready to ensure that I don't run into problems. Also to aid my workflow in mastering my composition.
I would firstly check to see if I have any polarity issues as I had learnt about this in my prior course on how polarity can affect audio. after checking this was fine I would load up my Loudness analyser by Youlean Loudness Meter, Next a tonal balance plugin which is a reference tool as it has references of many genres tonality and also the ability for me to see and hear the individual frequency contribution of each frequency ranges. I would also load my VSX headphone system and software for referencing my master in various environments and speakers.
Sometimes I start my mastering chain with a pre amp to either give coloration or saturation by harmonic content but In other scenarios i like to add an analogue tape reel emulation for a rich and warmer analogue style saturation. I have never used a real life analogue reel tape machine and have only studied on their use. I added the tape machine emulation from Slate digitals Virtual tape machine by Fabrice Gabriel (creator of EiosisEQ) and I would adjust the output higher (for more saturation) to a point in where I would hear a subtle difference in the change by the saturation introduced by the tape machine. (The difference is audible in the high and low end when I switched between the 2 tape speeds of the machine as demonstrated. I would next load the SSL Bus compressor next in my chain to shape and mould my tracks characteristics by taming the dynamics and "gluing the sounds altogether. After adjustments I would raise the level back up with the make up knob as I was hitting around 3 - 4db of gain reduction. I would check my loudness analyser to see how my loudness and dynamic range is of my track as too much compression can be bad for the dynamics of a track. I would also check my tonal balance to see how my track is comparing to other similar genres and also i would solo each frequency bands to hear the content in the selected band. I would continue in my mastering chain by loading Izotopes Ozone mastering plugin suite to help streamline my workflow. I would start with EQ and would listen to the lower frequency content. I would also listen to this in various listening environments such as a club, a car, small headphones, mobile phone, mono devices or far field speakers in a studio to better help my judgement and decisions in working with the low end so my track translates well in all environments. I would cut off frequencies below 40-50hz as I could hear this was only rumble and was just adding noise to my whole track. I have learnt in time not to completely cut off the low end as there are some tracks and genres which rely on the low end information and also some speaker systems
such as the Funktion - One Res 3 in clubs and large concerts or large ATC speakers. After adjusting the low end EQ I would do the same with the high end but would reference and listen to the audio in smaller speakers or headphones such as ear pods, mono devices and various headphones. I would next work on my kick and snare by firstly locating the frequency ranges and then boosting it up by a couple db. I would then begin working on the mid-side processing of my track to make my track sound wider by boosting the higher frequencies in the side channel where my drum overheads would be and other instruments such as guitars or keyboards . I would also cut the lower end so there it is less muddy in my stereo field and also because my lower end has been set to mono. After my changes I would solo the mids and sides to hear if I'm happy with my adjustments. After checking my loudness analyser, tonal balance and an A/B of before after I would move on to my next step of the mastering chain by adding an exciter which adds harmonic content in my track. This is so that I'm not just raising the gain/volume in frequencies where there isn't enough audio information in that specific area but instead raising it by adding colour and subtle harmonic content similar to what a tape machine does to my track but this exciter allows me to choose the type of harmonic saturation I would like on any frequency range I choose. I would set 4 bands (low, mid, high - mid, high) and would adjust them to fit each's respective frequency range and add very subtle amount of harmonic saturation as I already have a tape machine in my chain. I would also set the low end exciters saturation type to warm as I wanted a smooth low end but still audible on mono devices or small speakers. Next in my mastering chain is multiband dynamic processing and I don't use this heavily as its only used in my chain to help tame/ compress certain frequency bands for better control instead of a compressor which is more of a full band compressor. I would again create 4 bands and solo each to hear the frequency content in that selected area. If compression or dynamic control was required I would adjust the threshold and the mix bar which acts as a parallel mix between the wet and dry signal of my compression to a point I am happy with the overall sound of the track. Once again a task I do religiously is check my loudness analyser, tonal balance and A/B of before and after to see my progression and if I am working towards my reference/ target of master. To finalize my mastering chain, I used 2 limiters to elevate the overall volume of the final master to meet the targeted loudness level specified by streaming services or my clients' requirements. This ensures that the music maintains its clarity and integrity while adhering to industry standards for optimal playback quality. My reason of using 2 limiters is so that the 1st limiter catches and controls transient peaks while the 2nd is for overall loudness and a final polish. This multi stage approach also helps prevent distortion as the heavy load of limiting is split between 2 limiters. My 1st limiter I chose a vintage style limiter set to an analogue style with a slight fast attack time labelled "character" on the GUI . The 2nd limiter or maximiser would focus on maximizing loudness while preserving the dynamics of my track ensuring a polished final master.
After having a full listen thru and checking my integrated loudness (average loudness of my full track) and the overall dynamics of my track I was happy to see it fell along the lines of my targeted loudness
( -14 LUFS) which would be for major streaming services such as Spotify, Apple, Tidal and Amazon. To double check this as the standards could change or if my analyser was faulty I would upload my master to loudnesspenalty.com where I am able to check my masters loudness to the requirements of all major streaming platforms and also see and hear how much my master would be turned up or down if I was not near the recommended loudness of the streaming service.
With my track finally composed, mixed and mastered i would save a read out of the full analysis of the track with the loudness and dynamic detail for the client if required.
Summary
After a few listen throughs I was happy to say I had finished my task. Looking back and evaluating my practices and approach to this task I believe there was room for a lot of improvement on the instrumental side of my track as I had noticed I placed a lot of my instruments wide in the stereo field that they didn't sit well in the mix and the drums overpowered them. I also neglected some of the melodies in the 2nd half of my song and focused more on the drums due to my time management around this project. For future improvements I would ensure that I approach a task from a client with a more organised approach and to better arrange my timekeeping on a project. I feel in time as I develop a workflow or approach similar to mixing and mastering in the analogue domain I would be more streamlined in my workflow and better my mixing and mastering process with the addition of what I have learnt working in the digital realm. I also feel as I experience more genres and work in the industry I would better my understanding of the common mixing/mastering requirements as not every genre is mixed or mastered the same. For this project I had exhausted all my possible techniques in many areas of advanced production although in a normal scenario I wouldn't be excessive in using all these techniques or that many synthesizers. Although my master had met the loudness requirements of many distribution services I believe I hadn't gained the most potential possible from my master and I would do a 2nd revision mix and master.
